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Ixia Black Book: Voice over IP

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1. PR4 Unconfig IPv4 30 0 01 bo telooo01 Na Figure 177 Setting the eNodeB IP address b One MME with the IP address 30 0 1 1 is emulated by this configuration Select the IP MME element and set the Address and the Count hiii 11 Si v User Equipment vy IPRS Unconfig IPv4 16 0 0 0 1 0 0 0 0 Figure 178 Setting the MME IP address PN 915 2611 01 Rev H June 2014 193 Test Case Measuring Quality of Experience for Voice Calls in LTE 5 Configure the emulated User Equipment parameters a the APN Access Point Name must match the name configured on the SUT for this example the APN is set to aon 1 test com b the IP address of the PGW to which the APN refers must be set in the PGW IP for this case the real PGW has the IP address 22 0 0 1 Select the User Equipment element and in the Access Points tab set the APN name and the PGW IP Note The PGW IP address can be resolved by DNS query if the option Resolve DNS is enabled in the MME DNS tab in that case there is no need to fill the PGW IP field MME eNB 11 S1 U S IP NB IEE IP MME MAC ENB J MAcCmmME h ea triction Network provided APN subscription not verified Figure 179 User Equipment Settings Create the Network Traffic for Emulated P CSCF 6 Adda Net Traffic on the Terminate side Click on Terminate in the Navigation Pane or in the Terminate column in the Networks and Traffic pane then cloc
2. CPS Objective 100 calls per second Specify Talk Time 65000 ms 0 Specify Number OF Channels agpi Estimated Overhead Time i0000 ms Minimum Channel Inter call Duration 5000 ms Calculated Talk Time Figure 39 Configuration example for 100 cps with a test objective constraint of 8 000 channels Notes e Based on the values defined in the Custom Parameters tab and the test Objective Value IxLoad determines the call duration to be used at run time e The Estimated Overhead Time includes an estimation of the Call Setup Time and Call Teardown Time e The Minimum Channel Inter Call Duration enforces a minimum time between two calls that will be generated by the same phone this is required by some applications e In addition to the Talk Time option available as a configuration parameter on the RTP script objects the calculated talk time provided by the test objective is also available as a variable named TalkTime which is read only This variable can be used with the SLEEP script object which can accept it as a parameter for example SLEEP TalkTime allowing control of the call rate using configurations without RTP In this case the SLEEP script object replaces the Voice Session script objects used in this configuration PN 915 2611 01 Rev H June 2014 37 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring the Test Options 1 From the Test Configuration panel click Test Options
3. b UDP TCP TLS port Use same value c Phone no Use consecutive values per activity d Enable Accept multiple channels sharing the same IP port 6 Set the Channel mapping rules for RTP when using the same IP address for all streams a IP Address Use same value per port b UDP port Use consecutive values per port Configuring RTP Settings for the Receive_Call Activity Because all of the media streams will originate from the same IP address a range of RTP port numbers must be used Use the RTP tab to set a unique RTP port number for each one Remember that 8 000 RTP streams require 8 000 distinct port numbers with even values The odd port value is for RTCP traffic 1 2 Click the RTP tab Set the RTP Port to 10000 65534 2 The range starts from port 10 000 and increments by 4 until it reaches 65 534 A step of 2 will work for audio sessions For audio and video sessions a step of 4 is required PN 915 2611 01 Rev H June 2014 31 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems pal Plan RTP audio va Hardware acceleration Audiovideo port 10000 65534 2 _ ETEF Calculate advanced statistics Per Stream Statistics MDI Statistics Won blocking execution Figure 31 Activating the hardware RTP acceleration mode and RTP port configuration when all RTP streams are generated from a single IP address 3 Ensure that you have selected the Enable Hw Acc
4. First 100 callers disconnect the Last calls discon nected after 380 sec Active Calls calls here 5 min since the call 9000 started i All calls active for 220 sec 300 sec 80 sec 8000 l 7000 i n i 8 000 calls are connected in 80 seconds 6000 4 l l 5000 i 4000 i l l 3000 Last 100 callers disconnect the calls here 5 2000 4 min since their calls were connected 1000 i i 1 i 1000 0 50 100 150 200 250 300 350 400 450 Figure 42 Max active calls in time PN 915 2611 01 Rev H June 2014 50 Test Case Determining the Maximum Number of Concurrent Calls Setup The configuration simulates two H 323 networks each hosting 8 000 IP phones The calls will send voice traffic bi directionally and will have a constant call hold time of 5 minutes This example uses a single pair of Acceleron XP ports In this example we will assume that the phones in Network1 originate the calls and the phones in Network2 receive the calls and then immediately answer them The next figure highlights the test setup Network 2 919 5XX 1NNN H323 Phones H323 Phones Figure 43 Test setup PN 915 2611 01 Rev H June 2014 51 Test Case Determining the Maximum Number of Concurrent Calls Step by Step Instructions The final IxLoad configuration as a result of these steps is provided on the blackbook ixiacom com Web site see IxLoad 5 10 Voice H323 Concurrent Calls crf To import a Compres
5. Use Tel URI parameters phone context example com Phone book Edit Verify all settings Restore defaults Figure 100 Emulated endpoints dial plan 7 Set the authorization credentials of the emulated SIP endpoints This step is needed only for authorization enabled use cases In the SIP property page of the emulated endpoints the Originate network traffic define the username and password sequences under Authentication UAC For back to back setups in which IxLoad simulates both the endpoints and the registrar server the actual values of these fields are not important because the emulated Registrar server PN 915 2611 01 Rev H June 2014 111 Test Case Determining the Maximum Transaction Rate for VoIP Protocols does not validate them When the DUT is a real Registrar server however the credentials must match the settings on the server or the registration will fail Scenario Execution Dial Plan SIP Automatic TLS Cloud Codecs RTP Audio Video Fax T 38 J Enable signaling on this activity SIP Port soso if unchecked all SIP script Functions willbe SKIPPED JY Use external server m Transport settings Server address 20 1 100 1 Maximum message size on UDP hoz Server port 5060 R o f Domain name or local IP 20 1 100 1 Override transport specified in scenario UDP Only Outbound proxy TCP send immediate V Registrar server i 7 Auto register sim
6. pe gr ee ey BR ay eg ee a em gl l Private Network Internal I Public Network External il NAT NAPT x SRCIP 75 83 202 16 75 83 202 16 22 SRC PORT 32 000 192 168 1 100 24 SRC IP 192 168 1 100 SRC PORT 32 000 DSTIP 209 132 176 30 Re DSTIP 209 132 176 30 DST PORT 80 192 168 1 1 24 1 DST PORT 80 Figure 114 Outgoing HTTP request The incoming response from the HTTP server uses an IP address of 209 132 176 30 and port 80 Because the HTTP server received the packet from the public address of the NAPT enabled router 75 83 202 16 using port 32 000 it will reply back to this address When the router receives the response packet it will strip off the destination IP address and replace it with the host address 192 168 1 100 after performing a lookup operation in the existing entries of the NAT table The same lookup operation retrieves the destination port as well In this case it remains unchanged HTTP Client 192 168 1 100 32 000 receives an incoming connection with the response from the external HTTP Server 209 132 176 30 80 Private Network Internal l Public Network External Il NAT NAPT 192 168 1 100 24 DST IP 192 168 1 100 DST PORT 32 000 l l l SRC IP 209 132 176 30 75 83 202 16 22 ae PORT a DST IP 75 83 202 16 SRCIP 209 132 176 30 192 168 1 1 24 1 DST PORT 32 000
7. per second Registration completed How do the Registration Initiated and perseo SS Registration Completed rates compare A constant loop rate does not necessarily mean that the registration rate was maintained some of the loop can have failed For example the server can answer with a 500 Internal Error or 401 Unauthorized The 500 response is an indication in the Registrar server s database or a misconfiguration of one of the SIP Register message parameters The 401 response is an indication of a credentials mismatch PN 915 2611 01 Rev H June 2014 115 Test Case Determining the Maximum Transaction Rate for VoIP Protocols 3 To determine if the SUT handled all SIP message properly check the SIP Messages view SIP Messages VOIPSip 4 gt IEJ Drill Down AA E gt Stat Name 2 20 2 22 2 24 2 26 2 28 2 30 2 32 SIP VoIPSip REGISTER Requests Matched 97 611 99 103 100 582 102 071 103 593 105 126 106 597 SIP VoIPSip REGISTER Requests Parsed 97 611 99 103 100 562 102 071 103 593 105 126 106 597 SIP VoIPSip 1xx Responses Sent 97 611 99 103 100 582 102 071 103 593 105 126 106 597 SIP VoIPSip Requests Matched 97 611 99 103 100 582 102 071 103 593 105 126 106 597 SIP VoIPSip Requests Parsed 97 611 99 103 100 552 102 071 103 593 105 126 106 597 SIP VoIPSip 1xx Responses Matched 97 611 99 103 100 562 102 071 103 593 105 126 106 597 SIP VoIPSip 1xx Responses Parsed 97 611 99 103 100 582 102 071
8. SIP Tol From SIP From From SIP Tol From SIP Vial To SIP To To SIP_ Fron To SIP Vial To S5IP Contact Call ID AUTO CALL ID CSeq AUTO CSEQ Contact S5IP Contact Contact SIP Contact Contact SIP Tol Contact S5IP Contact Contact SIP From Contact SIP Contact Contact SIP Vial Max Forwards TO Content Length AUTO CONTENT LENGTH Expires 3600 Create From Template Load From File Edit Options Change c Modified by user W Case sensitive Delay before execution Static Expression lo ms Message body l Send audio SDP Offer Send custom message body Edit Custom Send a sip request message Restore Defaults Figure 232 Configuring multiple headers PN 915 2611 01 Rev H June 2014 248 Test Case Telephony Denial of Service 13 To enable the automatic processing of the configured variables in the Properties window select the Extract Variables tab and enable the checkboxes for the automatic variables used Consult the example below as reference Figure 233 PN 915 2611 01 Rev H Params Behavior Flow Manager Extract Variables Authentication Output _4 d Extract Variables gO SIP_Callld O SIP_Message O SIP_ReplacesParam q SIP Request Uni fel SIP_ From O SF Fom O SIP_FromTag O SIP_ToUn gO SIP_ToTag gO SIP_Cseq O SIP_CseqNumeric Value O SIP_CseqMethod SIP_Via H Vial ml SIP_Contact O SIP_
9. Stack 1 Insert below Ermulated Router Add above rat Delete O 1 1 1 Rename Masl em Import Overwrite Export Figure 12 Network1 Add an emulated router This will put the simulated phones behind an emulated router thus exposing a single MAC address for all the traffic The emulated router performs the same functions for the simulated subnet s that a real router performs for real subnets it forwards the packets between networks To configure the settings of the emulated router select the Emulated Router stack Each Ixia port you use in the test requires its own Emulated Router Configuring the Network Parameters for Network2 Note This network will host one SIP proxy server and one large media gateway RIP proxy which routes the traffic to a group of 8 000 phones that they serve The SIP proxy server will handle the SIP traffic and the media gateway will handle the RTP traffic To separate the signaling IP address of the SIP proxy from the media gateway address two separate network PN 915 2611 01 Rev H June 2014 18 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems ranges are required one for SIP and one for RTP The traffic configuration step will show how to associate network ranges with the SIP traffic or with the RTP traffic 1 Select Network2 this will display the IP Network Ranges 2 Add a secondary IP network range by clicking Add Row 3 Se
10. Table 44 Parameter Name Current Value Additional Options NAT type translation type User defined Sy Number of public IP addresses and port numbers User defined Number of private IP addresses and Userdeined a port numbers Results Analysis The DUT shall be monitored for e Memory size memory allocation de allocation issues while o Translations are added deleted to from the NAT table o Translations are continuously added and sessions are kept active PN 915 2611 01 Rev H June 2014 150 Test Case Using VoIP to Measure NAT PAT Performance e CPU usage e Size of NAT table The following questions provide guidelines on how to recognize specific problems during or at the end of the test execution 1 Has the test objective been achieved Check the Call Rates view Table 45 Call Rate statistics Statistic Name Questions Have the calls been attempted continuously at a constant call rate during the Sustain Time Calls Attempted per Second Calls Connected per Second How do the Calls Attempted rate and the Calls Connected rate compare to each other 2 Have any call failures been reported Check the Calls view Table 46 Call statistics Statistic Name Value Questions Calls Attempted a Calle Connected 1 Have any call attempts failed Calls Received Re a Calls Attempted and Calls Received Calls Answered ee en End Calls Received ay b Calls Attempted and Calls Connected End Calls Completed e
11. 4 436 3 391 3585 3635 3 668 FemaleMale_Mix3 35dBm 4 397 3 375 3 622 3 600 3685 25dBm 4 477 3 39 3723 3746 3701 Male1_Seq1 35dBm 4 449 3431 3701 3599 3 628 PN 915 2611 01 Rev H June 2014 97 Test Case Subjective Quality of Voice No degradations If the DUT performed ideally and did not induce any degradation the results will show a PESQ LE of 3 572 in the QoV PESQ view and a MOS of 4 72 in the RTP MOS view as shown in the following figure Onl PESO YoIPSip t ih IE RTF MOS voIPSip aa oh E Wh i Drag a column header here to group a ae Stat Name RTF oIPSip Stat Name RTP oIPSip RTF VoIPSip MOS Instant Avg aeee na aee na a een ac RTP volPSip RTP VoIPSip MOS Instant Best 4 190 4 130 4 130 4 130 4 RTP oIPSip RTF oIPSip MOS Instant Worst 4 130 4 130 4 130 4 130 4 RTP VoIPSip RTP WoIPSip MOS Best 4 130 4 130 4 130 4 130 4 RTP VoIPsip Drag a column header here to grou PESO LO Instant Maxi 3 571 3 571 3 571 3 571 PESO LG Instant Avg 3 571 3 571 3 571 3 571 IRTP VOIPSipy MOS worst 4 120 4 120 4 120 4 120 RTP volPSip MOS Per Call Avg 4 150 4 120 4 120 4 120 4 RTP WolPSip PESQ LE Max RTP voIPSin MOS Per Call Best 4 150 4 130 4 130 4 130 4 RTP voIPSip PESQ LE Avg JWRTP YoIPSip PESO LE Mind a 5l2 dol
12. 8 000 20 111 8 0 0 0 1 0 0 0 0 0 0 0 0 gee ee ae gt GratARP 1 Ethernet 1 Unconfigured Figure 45 Configuration example for Network 1 Configuring the Network Parameters for Network2 Note This network will host 8 000 H 323 IP Phones every simulated phone will use a unique IPv4 address for both H 323 and RIP traffic 1 Click Network2 to display IP Network Ranges PN 915 2611 01 Rev H June 2014 53 Test Case Determining the Maximum Number of Concurrent Calls 2 Set the following parameters IP Type IPv4 Hosts 8 000 First IP Subnet 20 2 1 7 Mask 8 Increment 0 0 0 1 Gateway 0 0 0 0 Gateway Step Size None 0229 5 p Q Table 15 Summary of Network 2 parameters IP Count Address Mask Increment Gateway Gateway MSS RX Type Increment 8 000 20 2 1 1 8 0 0 0 1 0 0 0 0 0 0 0 0 1460 Network a 2 3m MAGVLAN 2 E Ethernet 2 Sutogenerate Ma IP R2 Unconfigured IPv4 0 2 1 1 S000 0 0 0 0 0 0 0 0 Increment every subnet Figure 46 Network2 configuration example Configuring the Talk Time of 5 Minutes 300 sec The number of active calls depends directly on the call hold time talk time which is configurable within the test scenario or within the Activity settings the Audio tab We recommend you to use the settings at the activity level in this way the call duration can be changed from the automation tools because the parameters of th
13. E MediaGateway Consecutive IPs Network Range 2 in Networks 20 200 1 1 1 Figure 15 Configuration example for Traffic2 IP mappings 1 Select Traffic2 from Network2 2 Click the IP Mappings tab 3 For Group a Edit the name of the IP group from Group1 to SIPProxy and check that the group has Consecutive IPs rule set right click the IP Group Edit Group PN 915 2611 01 Rev H June 2014 20 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 42 Modify distribution group Group name SIPProxy amp IP 27 Groupi Consecutive IPs Distribution Create New Group Type Consecutive IPs 4 SGI croy Description Ewen distribution of IPs across ports For eg Network Rang Delete Group distribution of IP Network range 199 15 1 1 4 across 2 ports will be performed as Follows Porti will contain Mowe Range Up 198 16 1 1 2 and Port will contain 199 18 1 3 42 FA ye el r Move Range Down Cancel Figure 16 Modify the IP distribution group parameters b Close the Modify distribution group by clicking OK 4 Create a new Distribution Group Activities amp Endpoints Network Ranges By Port Distribution Group Receive Call Network Range IP R2 in Network2 20 1 100 1 13 Network Range IP R3 in Network2 20 1 200 1 13 6 amp dE a Figure 17 Create a new distribution list 5 For the new created distribution group DistGroupt1 a Edit the name of the IP group from DistGro
14. Eh Make _Cal MNetworki Calls Initiated Fer Second Timeline 1 0000 05 45 LE Receive Call hetwork Calls Initiated Per Second Timeline 1 000 05 45 Timeline Custom Parameters Timeline Hame Timeline Basic Ramp Up Type Channels Interval Scaling type Hours Ramp Up value 100 Ramp Up Interval 0000 00 01 100 Ramp Up Time 0000 00 04 Sustain Time 0000 05 00 a0 Ramp Down Value Ramp Down Time 0000 00 40 Iteration Time 0000 05 45 40 Iterations Time to First Iteration 0000 00 00 a Iterations Time Between Iterations ane 00 00 00033 O 01 06 0 01 39 Oziz 00z 45 0 03 18 003S Ogiz 00457 0 05 45 Figure 38 Configuration example for 100 calls per second 8 Under Network Traffic Mapping select VolPLink1 and then click the Custom Parameters tab PN 915 2611 01 Rev H June 2014 36 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 9 Within the CPS Objective parameters group select a Specify Number of channels 8 000 b Estimated Overhead Time 70 000 ms c Minimum Channel Inter Call Duration 5 000 ms Timeline and Objective Network Traffic Mapping Objective Type Objective Value Timeline z Mew Traffic Flow BM activity Links dA s Calls Initiated Per Second 100 Timeline 7 Make _Call Network1 Calls Initiated Per Second 100 Timeline 1 lB Receive _Call Network Calls Initiated Per Second 100 Timeline 1 Timeline Custom Parameters Bulk Call Generation Settings
15. Figure 60 H323 Receive_Call Configuration example for the Audio Settings page PN 915 2611 01 Rev H June 2014 64 Test Case Determining the Maximum Number of Concurrent Calls Configuring the Timeline amp Objective 1 2 Click Timeline amp Objective from the Test Configuration panel Set the test Objective Type to Channels Set the test Objective Value to 8 000 Under Timeline set the Ramp Up Value to 700 Set the Ramp Up Interval to 7 second Set the Sustain Time to 6 minutes Note The Sustain Time must be higher than the Call Hold Time It is also good practice to add some extra time for example 30 seconds as a buffer for any delays that may occur during the call setup and end call phases Hence ramp up should have a value higher than 300 sec 30 sec 330 sec 5 30 min Set the Ramp Down Time to 30 seconds Timeline and Objective Network Traffic Mapping Objective Type Objective Value Iteration Time Total T Mew Traffic Flow Activity Links Eh volPLinki Channels 6 000 Timeline1 0000 07 50 j BF Make_CalimNetwork Channels 6 000 Timeline 1 o00 07 50 Receive Call Network Channels 6 000 Timeline 1 0000 07 50 Timeline Custom Parameters Timeline Hame Timeline1 Basic Ramp Up Type Channels Interval Ramp Up Value Scaling type Hours Ramp Up Interval 0000 00 01 s000 Ramp Up Time 0000 01 20 7000 Sustain Time 0000 06 00 6000 Ramp Down value S000 Ramp Down Time 0000 00 30
16. PN 915 2611 01 Rev H June 2014 207 Test Case Measuring Quality of Experience for Voice Calls in LTE c Set the Audio Codec to AMR WB Select the Codecs tab in the table of audio codecs select the first one and choose AMR WB codec from the drop down list You have the option to change the order of preferred codecs The SDP will be automatically built with the parameters configured in this page 5 Cou Codecs BRP Audi Video Fax 738 Fax 7 30 SRTP MSRP Js Other Codec mode 0 6 60 kbps Incoming payload type 99 a ee Octet aligned format Outgoing payload type 99 _ Bandwidth efficient format lt i WARNING The outgoing payload type number ae ey ee I me 1 ic aa aaa will be used for RTP header payload type when Packet time eee ee sending packets and viceversa l 7 Please note that the outgoing payload type of a _ Enable VAD sender must be equal to the incoming payload AMR WB Adaptive Multi Rate WideBand is a multi mode type of a receiver codec that supports 9 speech encoding modes with bit rate between 6 60 and 23 85 Kbps 16000 Hz sampling frequency 50 fps Mode Request _ Change bitrate to 0 6 90 kbps after 0 packets _ Change bitrate to 0 6 90 kbps after 0 packets _ Iterate for the entire call duration _ Respond on Mode Change requests Figure 198 Set the AMR WEB as preferred codec PN 915 2611 01 Rev H June 2014 208 Test Case Measuring Quality of Experience for Voice Cal
17. In this example the proxy is configured to handle 1 000 calls while the Caller Callee Activity Link has a test objective of 100 Calls per Second with a maximum of 1 000 concurrent calls For 1 000 calls there are needed 2 000 endpoints 1 000 endpoints originating and 1 000 terminating calls In this configuration as in the majority of the IxLoad Voice configurations the Objective Type Channels is equivalent with endpoint 1 Click Timeline amp Objective from the test configuration panel 2 For the VoIPLink2 set the test Objective Type to Channels 3 Set the test Objective Value to 7 000 4 On the Timeline tab set the Ramp Up Value to 7 000 This means all the resources on the emulated SIP Registrar and Proxy server become active in the first second of the test execution 5 Set the Sustain Time to 1 hour The Sustain Time of this activity should be longer or at least equal with the sustain time of the Caller Callee Activity Link to ensure that all the registration requests and calls initialed by the emulated SIP User Agents are properly answered by the server 6 For the VoIPLink1 set the test objective to Loops Initiated Per Second For this call flow this is equivalent with Calls per Second while the test scenario contains one call 7 Set the test Objective Value to 7 000 8 Onthe Timeline tab set the Ramp Up Value to 700 PN 915 2611 01 Rev H June 2014 182 Test Case Determining the Performance of a Session Border
18. Voice over IP Contents How t Read this Oc Saga Sects selene as detente anal a ERE Fn dnsetnecueuta et aneennecueeitnanseenene Vil DIYITE viii VOI VOCE OVC H oec E a E a a a a r e E 1 Test Case Determining the Maximum Call Setup Rate CPS ccceeceeseeeeeeeeeeeeeeeeeeeeeeeeeees 7 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 11 Test Case Determining the Maximum Number of Concurrent CalllS ccccceecceeeeeeeeeseeeees 47 Test Case VoIP Quality of Service in Converged Networks cccsccccseseeceeeeeseeeeseeeesseeeesaaees 75 Test Case Subjective Quality Of VOICE 2 0 0 cccceecccesccccseeeceeeeeseeeeceeetaeeeeeeueetsueeeseneetsneeeseeeensness 87 Test Case Determining the Maximum Transaction Rate for VoIP Protocols secceeees 105 Test Case Using VoIP to Measure NAT PAT Performance cccceccceeeeseeeeeeeteneeeeeetaneess 119 Test Case Determining the Capacity of a VoIP to PSTN Gateway cccccccsececeeeeeeeeeeeeees 155 Test Case Determining the Performance of a Session Border Controller c sccceeeeees 173 Test Case Measuring Quality of Experience for Voice Calls in LTE ccceeceeeeeeeeeeeeeees 189 Test Case Measuring Quality of Experience for Multimedia VoIP Calls ccceeeeseeeeeeeees 213 Test Case Telephony Denial Of Service ccccecccecccceceteeeeceeeeneeteeeeeeeeteneesse
19. module in IxLoad PN 915 2611 01 Rev H June 2014 203 Test Case Measuring Quality of Experience for Voice Calls in LTE In the VoIPSIPCloud1 activity select the Diameter tab and check the Enable Rx Interface option Networks and Traffic TrafficFlow 1 g pina if E e O L lt gt Traffic2 VoIPSipCloud 1 VoIPSipCloud A Preview Cloud Traffic SeGrity Diameter Enable Rx Interface Diameter settings PCRF Transaction timeout ms 30000 IP or Hostname pecrf test com Watchdog timer ms 60000 Port 3868 Realm test com User defined AVP s Subscribe to notifications of signaling path status Provision signaling flow information Initial provision of session information Figure 193 Enable the Rx interface for the emulated P CSCF c Set the PCRF parameters The P CSCF needs to communicate with the PCRF the IP address of the Hostname and the realm of the PCRF are parameters of the Diameter configuration page In this example the PCRF hostname is pcrf test com and the realm is test com these are configuration parameters of the SUT Networks and Traffic TrafficFlow 1 v g 1 worki f w e Q e lt gt Traffic2 VoIPSipCloud 1 VoIPSipCloud Diameter Preview Cloud Traffic Enable Rx Interface Diameter settings PCRF Transaction timeout ms 30000 f test Er Hosiname pat test com Watchdog timer ms 60000 Port 3868 Rea
20. the IP address of the Registrar server may be entered in the Overwrite registrar field IxLoad has the ability to automatically send registration requests at the beginning of the test this simplifies the call flow allowing the user to concentrate on the call feature under test For a registration test case the automated behavior should be disabled Auto register simulated endpoints must remain cleared PN 915 2611 01 Rev H June 2014 112 Test Case Determining the Maximum Transaction Rate for VoIP Protocols 9 Define the Dial Plan and destination IP on the emulated SIP Registrar Terminate network traffic The SIP Cloud activity uses the information in incoming SIP Register Requests to route the message to the proper resource channel of the SIPVolPPeer activity By default all incoming requests are dispatched according to one of the default rules applied in the following order a Phone number in the To header must match the phone number in the Dial Plan source of this activity b Phone number in the From header must match the phone number in the Dial Plan Destination of this activity The To header of SIP Register request messages contain the phone number of the endpoint initiating the registration This is different than in other SIP request messages where the To header contains the information of the destination endpoint The Source dial plan on the emulated SIP Registrar activity should be configured identically to the dial p
21. 103 593 105 126 106 597 SIP VoIPSip 4xx Responses Sent 49 363 50 104 50 902 51 607 52 308 53 043 53 769 SIP VoIPSip 2xx Responses Sent 48 247 49 000 49 680 50 464 51 285 52 083 52 828 SIP VoIPSip Responses Orphans 0 0 0 0 0 0 0 SIP VoIPSip Requests Orphans 0 0 0 0 0 0 0 SIP VoIPSip Ignored Retransmissions 0 0 0 0 0 0 0 SIP VoIPSip Retransmitted Msgs 0 0 0 0 0 0 0 Figure 105 SIP messages view The SIP Messages view provides information about the number of SIP messages sent and received including retransmissions The values shown in this view should be consistent with the call flow In this example two Registration Messages were sent two 700 responses one 200 response and one 407 That means that the number of Register requests and 100 responses should be double the number of 200 and 401 responses at a given time Due to the statistics sampling interval every 2 seconds in this case it is possible that small differences will be seen For example the number of 200 and 407 responses may not be equal but they should balance by the end of the test Other statistics that should be checked include the following e Un dispatched SIP Messages under SIP Cloud stats if the number of un dispatched messages is greater than zero then the test has been misconfigured possibly because the dial plan of the emulated endpoints and the emulated SIP Registrar are not similar e VolP SIP Errors no errors should occur Test
22. 130 RTF ivoIPSip MOS Per Call Worst 4 110 4 110 4 110 4 110 4 110 4 110 4 110 Figure 87 MOS values in case of transcoding errors PN 915 2611 01 Rev H June 2014 101 Although the MOS score is perfect the PESQ LE score is 3 063 which is significantly lower Test Case Subjective Quality of Voice than the expected 3 512 Qoy PESO oIPSip Jh iat l H Drag a column header here to group Stat Name RTP VoIPSip PESQ LE Instant Min 3 063 3 063 3 063 3 063 3 063 3 063 RTF VoIPSip PESQ LE Instant Max 3 979 3 975 3 975 3 975 3 975 3 975 RTF VoIPSip PESQ LE Instant 4vq 3 519 3 463 3 546 3 463 3 519 3 519 RTF VoIPSip PESQ LQ Instant Ming 2 916 2 916 2 916 2 916 2 916 2 916 RTP VoIPSip PESO LQ Instant Max 4 133 4 128 4 128 4 128 4 128 4 128 RTF VoIPSip PESO LQ Instant Avg 3 522 3 447 3 557 3 448 3 525 3 523 RTP VoIPSip PESQ LE Min 3 063 3 063 3 063 3 063 3 063 B 063 RTP VoIPSip PESQ LE Max 3 979 3 979 3 979 3 979 3 979 3 979 RTF VoIPSip PESQ LE Avg 3 418 3 446 3 496 3 437 3 504 3 506 RIP VoIPSip PESQ LO Min 2 916 2 916 2 916 2 916 2 916 2 916 RTP VoIPSip PESO LO Max 4 133 4 133 4 133 4 133 4 133 4 133 RIP VoIPSip PESQ LQ Awg 3 368 3 426 3 492 3 479 3 502 3 505 Figure 88 PESQ values in case of Transcoding errors The source of this degradation is a malfunction of the DUT s transcoding module under stress conditions
23. 202 16 15000 Because the DBE SBC component stays in the middle of the media flow acting as the actual destination for the RTP packets initiated by Alice s phone it can provide both source and destination IP port information It uses this information to redirect the media packets from SBC to the public address of the CE NAT gateway 75 83 202 16 15000 rather than sending the RIP packets directly to the private IP address and port number received in the 2 INVITE SDP message 192 168 1 100 10000 Hence when the DBE SBC sends the RTP packets to 75 83 202 16 15000 the NAT will rewrite the destination IP address and port numbers to 192 168 1 100 10000 and the RTP packets will be successfully received by Alice s phone Objective This test determines the maximum number of VoIP video calls that can be established through a DUT when NAT is enabled The test methodology Determining the Max Call Setup Rate for SIP Based Devices and Systems uses voice traffic to determine the maximum number of concurrent calls that can be established through a device with or without NAT enabled VoIP video traffic and video conferencing support requires IxLoad 5 00 or later version In this test methodology the configuration is extended to generate voice and video traffic The video traffic adds more complexity to the NAT traversal because a video connection requires four media ports RTP audio RTCP audio RTP video and RI CP video compared with only two fo
24. 244 5060 SRC 75 83 202 16 5060 r 1 INVITE SDP as DST 193 16 148 244 5060 Alice 192 168 1 100 E2 2 INVITE SDP Fy Enterprise Session Border Controller NAT NAPT Nh ii 193 16 148 244 6 200 OK SDP 75 83 202 16 5060 SRC 192 168 1 1 5060 1025 5 200 OK SDP DST 192 168 1 100 5060 SRC 193 16 148 244 5060 DST 75 83 202 16 1025 Chas cc an an an am an an an cs ce ccc ae oae ae cc cs cs ae ces a as am ec ce cs a ee ce ee e ae a a ee ao ol INVITE SDP c IN IP4 192 168 1 100 m audio 10000 INVITE SDP c IN IP4 192 168 1 100 m audio 10000 200 OK SDP c IN IP4 193 16 148 245 m audio 20000 193 16 148 245 20000 200 OK SDP c IN IP4 193 16 148 245 m audio 20000 75 83 202 16 15000 192 168 1 100 10000 Figure 139 Media establishment between SBC DBE public interface and Alice s phone PN 915 2611 01 Rev H June 2014 144 Test Case Using VoIP to Measure NAT PAT Performance Finally Alice s phone processes the 6 200 OK SDP message using the IP address and port numbers received in the c and m lines of the SDP as a destination for the media packets After the ACK response is sent back to the SBC the conversation between Alice s phone and the external PSTN phone will be anchored by the DBE SBC component The first RTP packet generated by Alice s phone will open a new pinhole on the CE NAT gateway 75 83
25. 3 A VoIP user may be either the call originator or call terminator regardless of his location on a private network or public network 4 A VolP call uses even port numbers for RTP traffic and next available odd number for RTCP deviation from this rule will break the protocol conformance with RFC 3550 We will use a SIP example to better understand why those characteristics represent a challenge to NAT ed networks The same concepts can be extended to all VoIP protocols In this example we will use a simplified service provider network diagram Alice Enterprise Custome 193 16 148 244 17219 1001 Alice Enterprise Customer 193 16 148 244 NAT INAPT 192 168 1 100 NAT NAPT 192 168 1 1 24 4 o 172 19 100 100 172 19 100 2 40 244 220783 EOE EEEE E E EE f _ Service Provider Figure 119 VolP NAT Traversal PN 915 2611 01 Rev H June 2014 131 Test Case Using VoIP to Measure NAT PAT Performance The IP phones in the enterprise network use the 192 168 1 x private address range and are placed behind a NAT NAPT device that translates the private addresses to a single external address 75 83 202 16 The softswitch and media gateway located in service provider s network are also placed behind a NAT NAPT device that translates the 172 19 100 x private address range to an external address 193 16 148 244 The IP phones from the enterprise network are configured to use 193 16 148 244 5060 as their outbound proxy as instruc
26. 48 Test Case Determining the Maximum Number of Concurrent Calls Determining the Maximum Number of Concurrent Calls for H 323 Based Devices and Systems Overview The goal of this test methodology is to help you determine the peak capacity measured in active calls supported by a DUT that uses the H 323 protocol The DUT may include one or more of H 323 application layer gateways SBCs H 323 gateways signaling gateways IP softswitches and IP PBXs While the primary metric is the number of calls that can be sustained by the DUT you must keep the user experience in the desired range Test Methodology Set the performance goals max active calls calls completion percentage post dial delay post pickup delay TCP retransmissions media delay MOS PESQ packet loss percentage jitter and one way delay Use a binary search to determine the maximum call rate using short calls with duration of at least 15 min 80 min recommended o Verify whether The calls connect without call failures The H 323 QoS meets expectations The RTP QoS meets expectations o Plot achart of active calls versus QoS o Plot achart of active calls versus DUT CPU utilization o Plot achart of active Calls versus DUT RAM utilization Repeat the test for every supported audio CODEC For every CODEC repeat the test for all packet times supported by the DUT The performance can differ dramatically between a ptime of 5 ms and a ptime of 30 ms After
27. Bole aol2 f idle nici aiid aicka a Role ole Sole 3 512 3 ra adel oie hair lo RTP voIPSip MOS Per Call worst 4 130 4 120 4 120 4 120 4 RTP WoIPSip PESO LO Min Figure 81 MOS and PESQ values for the ideal conditions The values for Lost Packets are zero as shown in the following RTP QoS view E RTP Qo5 volPSip EBA E4 t Eh Stat Name 40 42 44 46 48 l IRTP ivolPSip Packets Sent 2 209 591 2 450 663 2 652 926 2 815 752 2 939 538 AIP volPsip Packets Received ee 2 210 073 2 450 789 2 653 068 2 815 896 2 939 683 RTP VoIPSip Bytes Sent 07 235 204 107 829 172 116 728 744 123 893 088 129 539 672 RTF volFSip Bytes Received 07 243 212 107 834 716 116 734 992 123 899 424 129 346 052 RTP fvolPS5ip Throughput Outbound kbps 45 681 42 485 34 679 20 669 20 930 RTF VoIPSip Throughput Inbound kbps 45 678 42 480 34 680 28 670 20 936 RTF VoIPSip Tx Packets Dropped D D D D pf ROG le aa a RTP VWolPSip Maximum Consecutive Lost Packets a a a D D RTF VoIPSip Packet Errors Received 5 5 5 5 5i RTF volP5ip Duplicate Packets Received 46 46 46 46 46 Figure 82 RTP QoS in the case of No degradation PN 915 2611 01 Rev H June 2014 98 Test Case Subjective Quality of Voice Packet loss lf there are impairments at the network level such as packet loss delay and jitter the quality of voice reported by both metrics MOS and PESQ will be affected The results
28. Concurrent Calls 3 Wedo not recommend you to set the destination while this activity just receives calls and does not generate calls Traffic Receive _Call WoIPH323 Peer Source Destination IPs The source IP addresses are taken From the associated Network see Traffic Network mapping tables in the test IPs None Phone numbers Alias Override phone numbers from destination activity C Phone book entry User defined 919501 1000 Type oF Alias Dialed Digits Type of number National number Numbering plan ISDN telephony numbering Phone book Figure 56 Receive_Call Dial Plan configuration example for Network 2 Configuring the H 323 Settings for the Receive_Call Activity 1 Click the H 323 configuration page of Receive_Call 2 Verify that the Enable signaling on this activity check box is selected 3 Verify that the Enable FastStart check box is cleared 4 Verify that the Enable Tunneling check box is cleared 5 Set Q 931 User User to 918501 1000 6 Set Q 931 Display to xia 918501 1000 PN 915 2611 01 Rev H June 2014 62 Test Case Determining the Maximum Number of Concurrent Calls 7 Verify that the Enable RAS check box is cleared Traffic Receive Call VoIPHS23 Peer Dial Plan H323 Terminal Capability Enable signaling on this activity iF unchecked all H323 script Functions will be SKIPPED H323 Specific Settings Advanced Signalling Options Bandwidth 4nd T
29. Contact Centers A soft copy of each of the chapters of the books and the associated test configurations are available on Ixias Black Book website at http www ixiacom com blackbook Registration is required to access this section of the Web site At Ixia we know that the networking industry is constantly moving we aim to be your technology partner through these ebbs and flows We hope this Black Book series provides valuable insight into the evolution of our industry as it applies to test and measurement Keep testing hard A Errol Ginsberg Acting CEO PN 915 2611 01 Rev H June 2014 vill Voice over IP Voice over IP Test Methodologies This booklet provides baseline test execution plans for running common VolP testing scenarios It provides a common set of recommendations and guidelines for running the test cases against various VoIP implementations PN 915 2611 01 Rev H June 2014 Ix Voice over IP VoIP Voice over IP VoIP refers to the transmission of voice data over IP enabled networks rather than using the traditional circuits of the PSTN Public Switched Telephone Networks Today VoIP is associated with IP telephony which refers to the family of protocols used to deliver the voice data RTP SRIP H 323 SIP MGCP MEGACO H248 SCCP Skinny and SIGTRAN VoIP Benefits The most attractive benefit of VoIP is the cost savings which can be substantial in large enterprises and for service providers The savi
30. Controller 9 Setthe Sustain Time to 1 hour 10 Inthe Custom Parameters tab select the Specify Number of Channel check box and set its value to 7 000 Execute the Test Map the ports and run the test Test Variables Table 63 Parameter Name Current Value Additional Options IP Type Type of traffic IP Mapping rules for User agents Number of users available User defined o Number of active calls User defined o Call Rate User defined o Call Duration User defined G 711a G 729 G 723 G 726 iLBC AUDIO pe Packer seers G 711u AMR any other codec using Custom frequency Codec Mix with data protocols for Not included Any combination of data protocols example FTP HTTP Telnet supported by IxLoad successfully calls canceled calls Mix of call flows unanswered calls busy calls call forward call transferred call hold retrieve Mix of call features Results Analysis The DUT shall be monitored for the following e Memory size memory allocation de allocation issues while o Translations are added deleted to from the NAT table o Translations are continuously added and sessions are kept active e CPU usage e Size of NAT table The following questions provide guidelines on how to recognize specific problems during or at the end of the test execution 1 Has the test objective been achieved Check the Call Rates view PN 915 2611 01 Rev H June 2014 183 Test Case Determining the Performance of a
31. Eche Ff i oe H Badd Channel GarRemove EcCompact 3 Note zoom 100 Full Screen button PN 915 2611 01 Rev H June 2014 23 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 4 Locate the RTP Voice Session script objects used by Scenario Editor The scenario has two functions one on the caller side and one on the called side D H a ar sO xXlo m lace aaa ig mi Bc Add channel S Remove HE Compact A Note zoom 100 Scenario Editor SIP_MakeCall ReceiveCall EndCall with RTP Scenario Channel 0 Activity Make_Call Traffic Trafficl Network1 Column Originate Link VolPLink41 i pae Ses E is Recei aN SIP End ow Is Error a Ere Error Figure 21 Scenario Editor in full screen mode VOICE SESSION objects 5 For both Voice Session script objects included in the script do the following a Double click the Voice Session script object opening its properties b Inthe Talk Parameters tab clear the Overwrite playback activity settings check box c Click the Listen Parameters tab and then clear the Overwrite playback activity settings check box d Click OK to close the properties page e Remember to repeat these steps for the second voice session script object PN 915 2611 01 Rev H June 2014 24 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Yoice Session Properties ol Yo
32. Extension to Other Protocols Quality of voice analysis using PESQ is performed on RIP traffic and is independent of the call establishment protocol The same test methodology can be used for different signaling protocols The same methodology applies when different signaling protocols are used by each party for example H 323 on one side and SIP on the other PN 915 2611 01 Rev H June 2014 102 Test Case Subjective Quality of Voice Conclusions Quality of voice is the most important measurement of a voice communication system s QoS as experienced by users Two methods and metrics are commonly used MOS based on E Model and MOS based on PESQ E Model is computed using transmission metrics and associated constants to approximate the degradation induced by sampling and coding of the speech signal PESQ compares the audio signal received with the expected one PESQ requires much higher processing power than E Model but has the capability to detecting degradations that are not visible with E Model lf a DUT only operates at the packet level and does not affect the audio streams MOS scores based on the E Model will provide a good and complete characterization of the expected Quality of Voice If the DUT performs media operations such as AGC VAD decoding encoding an in depth analyze of QoV should be performed by the perceptual full reference PESQ method PN 915 2611 01 Rev H June 2014 103 Test Case Determining the Maximum Trans
33. Override default dispatching rules has the proper sequence of phone numbers in the Formula for dispatching field 55517 0000 in this example Dial Flan Cloud Weadaee Wit andia ides Wes ft 208 E et an enra Iarbas ft Cloud Rules Enable SIP Cloud simulation using settings From SIP SP ra ae Channel Dispatching Rules 1 Ti m 1 Ti ps m P im Larm te Cloud topology n sc 6551 0000 Mame sip_server l Simulat Close Dispatching rules Override default dispatching rules faal Figure 170 Cloud settings for R_Caller activity PN 915 2611 01 Rev H June 2014 180 Test Case Determining the Performance of a Session Border Controller Setting the R_Callee Activity Parameters The activity handling the registrations of SIP Callee user agents is R Callee 1 Click the R_Callee activity under SIP_Proxy NetTraffic 2 Check the Scenario it contains a procedure waiting for SIP Register and sending the response 200 Ok It can be modified to emulate the Register with authentication enabled by adding script objects functions to send 401 Unauthorized and wait for the second Register 3 Click the Dial Plan tab and set the Source Phone Numbers 5552 0000 in this example this sequence should match the one specified as Source Phone Numbers for the SIP_Caller activity this is the criteria for matching the incoming SIP Register messages to this activity 4 Click the Cloud tab and ensure that the Enable SIP Cl
34. PN 915 2611 01 Rev H June 2014 164 Test Case Determining the Capacity of a VoIP to PSTN Gateway Set Audio Parameters for PSTNDigitalPeer1 The Audio parameters the clip to be played and the played duration may be set in the test script objects or at the Activity level To control the audio parameters from the test script objects the Overwrite playback activity settings check box has to be selected in the Voice Session function or whatever media function is used It is preferable to use the settings at the Activity level to allow control of the audio parameters from the automation scripts 1 Click the Audio tab 2 Select the Play for check box 3 Set Play for to 50 seconds Traffic PSTNDigitalPeer1 PSTNDigital Peer Scenario Execution Settings Dial Plan Play settings Clip U5 O42 way hi Format PCM Duration 32785 ms Size 524556 bytes Play For clip duration or TalkTime all objectives except Channels i Play For 50 Seconds Verify all settings Restore defaults Figure 155 Audio settings for PSTNDigitalPeer Configuring the Timeline and Objective 1 Click Timeline amp Objective from the test configuration panel 2 Set the test Objective Type to Channels This is equivalent with concurrent active calls 3 Set the test Objective Value to 30 This value may be increased if multiple spans are available the number of IP addresses for SIP activity and the dial plan range have to be exten
35. Retransmitted Msgs statistic located under the SIP Messages view 5 Has the QoS for media met the expected quality Check the RTP MOS RTP QoS RTP Advanced QoS RTP Jitter Distribution RTP Consecutive Lost Datagram Distribution and RTP Streams statistic views Table 49 MOS statistics Statistic Name Questions How do the last values reported by the RTP MOS Best and RTP MOS Best RTP MOS Worst compare with each other RTP MOS Worst How does the RTP MOS Worst score compare with the max theoretical score for the CODEC used Are any times without an instantaneous MOS value How frequent are the changes in the instantaneous MOS values RTP MOS Instant Best Avg Worst RTP MOS Per Call Howdo the MOS per Call statistics compare with the RTP Best Avg Worst MOS Best and RTP MOS Worst statistics PN 915 2611 01 Rev H June 2014 152 Test Case Using VoIP to Measure NAT PAT Performance Table 50 Basic RTP QoS statistics see RTP QoS and RTP Advanced QoS statistics views Statistic Name Questions RTP Packets Sent 1 Are there any differences between RTP Packets Sent and RTP Packets Received RTP Packets Received RTP Packets Lost Does the difference match the value of RTP Lost Packets RTP One Way Delay us Is the One Way Delay higher than 100 ms RTP Delay Variation Jitter us RTP Interarrival Jitter us What is the max Delay Variation Jitter 5 What is the max Interarrival Jitter Table 51 RTP Jitter di
36. Send Request Properties Parameter Expression Sia Auto Variables gO AUTO CALLID S P_Callld O AUTO REPLACES PARAM SSIP_ReplacesParam O AUTO REQUEST URI 5 P_Request Uni gO AUTO FROM S IP_From O AUTO FROM LURI S1IP_FromUri gO AUTO FROM TAG S P_FromTag g AUTO TO SSIP_To g AUTO TO URI SIP_ToUn g AUTO TO TAG SIP_ToTag gO AUTO CSEQ SSIP_Cseq g AUTO CSEG NUMERICVALUE SIP_CsegNumenc O AUTO CSEQ METHOD S51P_CseqMethod gO AUTO VIA SSIP_Via gO AUTO VIA BRANCH S P_ViaBranch O AUTO CONTACT SIP_Contact gO AUTO CONTENT LENGTH 5 P_ContentLength gO AUTO ROUTE S5 P_Route Custom Behavior m Transport Layer Dialog Layer UDP Wo Newdialog Existing dialog Remove previously received messages Global W Extended variables support Send a sip request message Restore Defaults Cancel l Figure 229 Enabling the Extended Variables Support for the current procedure 8 Change the Contact header information for both Re Send REGISTRATION procedures and enable the checkbox for Extended Variables Support 9 To change the state machine execution and create a larger diversity of test cases re link the output points of the existing procedures to others and create different execution flows for the Registration process Depending on the DUT configuration the Registration may succeed or fail and no malfunctions or security breaches should be encountered As an example after a few unsuccessful registration attempts th
37. T1 04 Standalone Internal om Mok driven NIA TOB E1 T1 04 Standalone Span Mot driven hy AEN ax Eee Figure 158 Configuring the PSTN board clock setting 2 Set the Source Reference Clock to Internal The DUT is configured to synchronize its clock to the Span so both IxLoad and DUT are synchronized to the clock generated by IxLoad Running the Test 1 Save the configuration 2 Set Global settings optional step There are several parameters that control the PSTN interfaces behavior These parameters are usually not changed between devices under test Their values can be changed in the Global Settings that can be accessed by clicking Global Settings in the scenario tab of a VoIP or T1 E1 activity Traffic PSTMDigitalPeer1 PSTNDigital Peer Scenario Execution Settings Dial Plan Audio D awe W Ge cou BOB Kl o o HE j p mi B Add Channel Bw Remove HE Compact H Note eesse 6 workspace a Scenario Editor SIP PSTN basic call c Glob Setti ings Button oo E oh 4 i S u a Make Call f connected o Disconnected Digital only iby Make Call fiat Receive Call g End Call Voice JP ro Figure 159 PN 915 2611 01 Rev H June 2014 168 Test Case Determining the Capacity of a VoIP to PSTN Gateway Te Global Settings E ixload IxLoad xLoadt5 10 0 238 EBMxVoicePluginPoolWefault lst x5 Execution Settings Globa
38. The final IxLoad configuration as a result of these steps is provided on the blackbook ixiacom com Web site see IxLoad 5 10 Voice SIP Back to Back User Agent crf crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test PN 915 2611 01 Rev H June 2014 174 Test Case Determining the Performance of a Session Border Controller Setting the Network Parameters 1 Import the configuration IxLoad 5 10 Voice SIP Back to Back User Agent crf dh gt Originate db DUT an Terminate 2 2 SIP_Caller fy SIP_Prony B m ror ope SE Traffict db Traffic fd Caller Eal gP 2 Traffic3 T Talee Eal Figure 164 SIP Back to Back User Agent configuration 2 Click the SIP_Caller NetTraffic and change the network settings accordingly to the particular test topology The number of IP addresses should match the number of channels defined in Test Objective 1000 in this example 3 Click the SIP_Callee NetTraffic and change the network settings accordingly to the particular test topology The number of IP addresses should match the number of channels defined in Test Objective 1000 in this example 4 Click the SIP_Proxy NetTraffic and for IP R3 the first IP range change the net
39. Using VoIP to Measure NAT PAT Performance The proxy server replies with 200 OK SDP using the address and port 172 16 100 2 5060 from inside the IP and SIP headers For media the proxy server uses the media IP address and port number exposed by the SBC DBE media gateway 172 16 100 3 30000 Hence the SDP body attached to 4 200 OK SDP response includes c IN IP4 172 16 100 3 m 30 000 audio RTP AVP 0 101 This allows the SBC DBE component to generate and receive RTP messages between 172 16 100 10 25000 and 172 16 100 3 30000 On the public side of thecall as seen from SBC s point of view upon receipt of the 4 200 OK SDP response the SBE SBC component sends the 5 200 OK SDP response which includes the public media address of the DBE SBC component 193 16 148 245 20000 The message is sent to the public address of the CE NAT 75 83 202 16 1025 The SDP body of the response includes the following media lines c IN IP4 193 16 148 245 m 20 000 audio RTP AVP 0 18 101 Upon receipt of the 5 200 OK SDP response the CE NAT will initiate the reverse translation from its public address 75 83 202 16 1025 to the private address 192 168 1 100 5060 and the message will be sent toward Alice s phone without modifying the payload The CE NAT only swaps the destination IP address and port fields of the IP header This message is highlighted as message 6 200 OK SDP in the figure below SRC 192 168 1 100 5060 DST 193 16 148
40. VELT Page Channel mapping rules for SIP Use same value per port UA IP Address Channel mapping rules for SIP Use same value UA UDP TCP TLS port Channel mapping rules for SIP Use consecutive values per activity UA Phone no Channel mapping rules for RTP Use same value per port IP Address Channel mapping rules for Use consecutive values per port RTP UDP port Audio Video port 30000 65535 4 The same IP address is used for Video and VolP traffic so the UDP port range is shared between the IPTV and VoIP activities Enable audio PN 915 2611 01 Rev H June 2014 81 Test Case VoIP Quality of Service in Converged Networks 10 Set the test objective and timeline for each activity Table 34 Test objective Activity Objective Type Objective Value For all activities use the same Timeline Timeline 1 with Ramp Up value 70 and Sustain Time 70 00 minutes The objective values for each activity are set to meet the proposed throughput distribution between the different protocols With these values the total inbound throughput is 175 Mbps 11 Assign the ports to generate traffic at least four ports are required The test may be scaled up by increasing the number of ports and increasing the test objectives proportionally 12 Start the test and activate the following views e Video Client Data Rates e HTTP Client Throughput Objective e Calls VoIPSIP e RTP MOS VoIPSIP PN 915 261
41. a BYE request that will gracefully disconnect the call Calls without media may be reported by the test tool in the ramp down phase for calls initiated but not connected Trafficl Make_Call VoIPSip Peer Run For Loop delays Aliases the entire test duration Before 1st loop ms Number of aliases phone numbers per channel 1 C a number of loops Between loops 0 rs NOTE IF more than one aliases will cycle Channel mapping rules For SIP UA Channel mapping rules For media IF address Use consecutive values per pork IF address Use consecutive values per port UDF TCP TLS port Use same value Fort Use same value Phone no Use consecutive values per port Accept multiple channels sharing the same IP Port verify all settings Restore defaults _ Graceful Ramp down Figure 23 Configuration example for the Execution Settings tab of the SIP Make_Call activity 5 Channel mapping settings for SIP when simulating SIP IP Phones a IP Address Use Consecutive Values per port b UDP TCP TLS port Use same value c Phone Number Use Consecutive Values per activity 6 Channel mapping rules for RTP when simulating SIP IP Phones a IP Address Use Consecutive Values per port b UDP TCP TLS port Use same value Configuring the Dial Plan for the Make_Call Activity 1 Click the Dial Plan tab 2 Set Source Phone numbers by selecting the Specify Number URI check box 3 Set the source phone number by c
42. adapter card and appropriate number of E1 T1 modules Concurrent Calls Drecionotealis SIP to PSTN PSTN to SIP the call flow in test scenario and the dial plan have to be changed to run this test ISDN VN6 KHT KOR TWN AUS PSTN Protocol ISDN QSIG CAS MFC R2 Conclusions This configuration covered the main parameters of the SIP to PSTN configuration by using a practical example allowing the user to control the concurrent number of active calls running capacity tests The results section covered the main statistics that may highlight an issue PN 915 2611 01 Rev H June 2014 172 Test Case Determining the Performance of a Session Border Controller Test Case Determining the Performance of a Session Border Controller Overview The challenges and test methodology for Session Border Controllers and Application Layer Gateways testing were discussed in this Black Book in the section TEST CASE DETERMINING THE MAX CALL SETUP RATE FOR SIP BASED DEVICES AND SYSTEMS That use case is focused on originating calls from one network Private and terminate them on the other network Public In that case a single call leg is set for each call There are cases when the calls are established between two endpoints located in the same network while the SIP server is in a different network In these cases the DUT handles two call legs for each call one between the originator of the call and SIP proxy and second between SIP Proxy and the te
43. and ensure that the Enable signaling on this activity check box is selected Configure the originating SIP port as needed This example will use the 5060 default value Increase the value for Maximum message size on UDP to 2500 bytes This parameter controls the SIP buffer size for the transmitted packets Configure the desired user name and password for the authentication of the endpoint This example will use ixia for the User name and Password Enable the checkbox Use external server and configure the DUT address or name This example uses the following parameters 19 Table 87 Parameter Value 20 20 20 200 5060 Domain name or 20 20 20 200 local IP Outbound Proxy Checkbox een oeo Registrar server Checkbox eee oeo Enable the checkbox Override default destination domain name or host port and configure the destination IP or name of the DUT This value is used in the construction of the SIP To header for the transmitted packets Ignoring this value and leaving the default should construct PN 915 2611 01 Rev H June 2014 252 Test Case Telephony Denial of Service an improper message that the DUT should ignore Omitting this value or configuring other values represents a new test case This example will use the value 20 20 20 200 Scenario Execution aling on this activity if uncheckec 1 all SIP script functions will be SKIPPED Transport settings Maximum message size on UDP 2500 _ Ov
44. cannot be established the reasons for failure can be identified in the Event Viewer RTP MOS VoIPSip ete Tl oe e os e a sl el sl ee MOS Instant Avg MOS Instant Best 3 820 3 820 3 620 3 620 3 820 3 820 3 620 3 820 3 620 3 620 3 820 3 620 3 8620 3 820 3 820 MOS Instant Worst 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 MOS Best 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 MOS Worst 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 Oo 6l MOS Per Call Ava MOS Per Call Best 8 MOS Per Call Worst 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 820 3 82 3 820 3 820 3 620 3 820 3 820 3 620 3 820 3 620 3 620 3 820 3 620 3 820 Figure 202 Quality of Voice Quality of voice is the final goal of this test even though the calls can be established it is important to measure QoE for the voice service In this case the measure MOS is as expected for AMR WB codec mode 0 the expected MOS is 3 82 PN 915 2611 01 Rev H June 2014 211 Test Case Measuring Quality of Experience for Voice Calls in LTE Troubleshooting and Diagnostics When issues are present a deep down analysis can be done using various features provided by IxLoad Stats drilldown per activity port Event viewer for SIP SDP and RTP related issues wh
45. described in this chapter represents a configuration baseline and additional alterations can be performed to stress in multiple ways the stability and security of the system using other methods like INVITE OPTIONS SUBSRIBE ACK BYE or NOTIFY The most challenging part of the configuration is determining the settings that have the most impact on the DUT as a system failure during live operation can bring major penalties to the security of the voice infrastructure and business operations PN 915 2611 01 Rev H June 2014 206 Corporate Headquarters Ixia Worldwide Headquarters 26601 W Agoura Rd Calabasas CA 91302 USA 1877 FOR IXIA 877 367 4942 1818 871 1800 International FAX 1 818 871 1805 sales ixiacom com EMEA Ixia Technologies Europe Limited Clarion House Norreys Drive Maiden Head SL6 4FL United Kingdom 44 1628 408750 FAX 44 1628 639916 VAT No GB502006125 salesemea ixiacom com Ixia Asia Pacific Headquarters 21 Serangoon North Avenue 5 04 01 Singapore 5584864 65 6332 0125 FAX 65 6332 0127 Support Field Asia Pacific ixiacom com PN 915 2611 01 Rev H June 2014 Contact Ixia Web site www ixiacom com General info ixiacom com Investor Relations ir ixiacom com Training training ixiacom com Support support ixiacom com 1 877 367 4942 1 818 871 1800 Option 1 outside USA online support form http www ixiacom com support inquiry Renewals renewals emea ixiacom com Support sup
46. expected This is an indication of errors in the speech processing component of the DUT The RTP QoS statistics in the following table shows no loss errors or delay of the packets RIP O05 olPSip ity TRl Drag a column header here to grou 42 Stat Name 44 46 48 RTP oIPSip Packets Sent 1 728 291 1 804 749 1 841 960 1 848 000 RIP volPSip Packets Rec 1 725 451 1 801 949 1 899 185 1 848 000 RTP olPSip Bytes Sent 76 044 804 79 408 956 81 046 240 981 312 000 RIP WoIPSip Bytes Recei 75 919 844 79 285 756 80 924 140 81 312 000 RTP olPSip Throughput 20 344 12 286 6 549 525 RTP y olPSip Throughput 20 355 12 216 6 553 399 RTP VoIPSip Tx Packets 0 0 0 0 E C a CYC RTF voIPSip Maximum Zo 0 0 0 0 RTF voIPSip Packet Error 0 0 0 0 Figure 86 RTP QoS in case of transcoding errors Consequently the MOS score is perfect iv TE i Drag a column header here to grou Stat Name 42 i RTF VoIPSip MOS Instant Avg O 4 130 4 130 4 130 4 130 RTF VoIPSip MOS Instant Best O 4 130 4 130 4 130 4 150 RTF VolPSip MOS Instant Worst D 4 130 4 150 4 130 4 130 RIP VoIPSip MOS Best 4 130 4 130 4 130 4 130 4 130 4 130 4 130 RIP GolPSp MOS Worst palo 4110 ato 4110 aTh 4 110 RTP VoIPSip MOS Per Call Avg 4 120 4 120 4 120 4 120 4 120 4 120 4 120 RIP VoIPSip MOS Per Call Best 4 1350 4 130 4 130 4 130 4 130 4 1350 4
47. in this case show a minimum MOS score of 2 95 and a minimum PESQ LE score of 2 858 Under ideal conditions these values should be 4 72 and 3 512 respectively Qoy PESO YoIPSip Jh 4r LG RTP MOS VoIPSip Jh a ae fe Drag a column header here to ji o Stat Name RTP vWolPSip PESO LE Instant Min RTP VoIPSip PESO LE Instant Max RTP VoIPSip PESO LE Instant Avg RTP vWolPSip PESO LO Instant Min RTP WolPSip PESO LO Instant Max RTP WolPSip PESO LO Instant 4vq RTP volPSip PESO LE Max RTP volPSip PESO LE avg RTP volPSip PESO LO Min Drag a column header here to group Stat Name RTP VoIP Sip MOS Instant Avg RTP VoIP Sip MOS Instant Best RTP VoIPSip MOS Instant Worst 2 990 2 990 2 990 RTF VoIPSip MOS Best Tae a E RTP volPSip MOS Per Call vq 3 960 3 960 3 960 RTP VoIPSip MOS Per Call Best 4 130 4130 4 130 RTF VoIPSip MOS Per Call Worst 3 480 3 460 3 480 aidbar didlo dadla 444 agar a47 foie Aol elt 3571 35i 35l 4 130 4130 4130 2059 2 000 2 050 Si athe a S 3 455 rf 3475 Sao A a 2 014 m1 ril ril m1 mi mi m1 ri ri rs Figure 83 MOS and PESQ scores in case of Packet loss Poor scores for both metrics is an indication that the DUT has induced degradation in the packet flow To identify the type of degradation RTP QoS values can be used In the case discussed here there are lost
48. location of the pinhole MAPPED ADDRESS that was used to send the binding request so an application can use this information in its payload The SIP client will further use the extracted MAPPED ADDRESS to build the payload of the SIP requests due to the different destination of the SIP packets as the SIP server has a different address than the STUN server a new pinhole will be created by a symmetric NAT leading to the use of an invalid MAPPED ADDRESS information into the payload of the SIP request TURN Traversal Using Relay NAT TURN is a simple protocol that tries to resolve STUN s issue with symmetric NATs by allocating a public IP port on a public server that is used to relay media between communicating parties While TURN can almost always provide connectivity to a client it has the disadvantage of introducing a high cost to the service provider hosting the TURN server PN 915 2611 01 Rev H June 2014 136 Test Case Using VoIP to Measure NAT PAT Performance ICE Interactive Connectivity Establishment ICE describes a methodology for network address translator traversal for the session initiation protocol SIP ICE is a framework that combines existing protocols such as STUN TURN and real specific IP RSIP by choosing the best interconnection method ICE provides a NAT traversal solution that is independent from the types or number of NATs in use The advantages provided using ICE are as follows e It provides the
49. most economical solution for a service provider e It always results in the minimum voice latency e It can be done without any increase in the call setup delay e Itfacilitates the transition of the Internet from IPv4 to IPv6 supporting calls between dual IPv4 IPv6 stack clients and IPv6 clients behind an IPv4 to IPv6 NAT device e lt uses SIP preconditions to guarantee that the phone does not ring unless the users will both hear and see each other when they pick up The disadvantages are e ICE is not yet standardized e ICE is designed only to work with SIP Application Layer Gateway ALG The ALG is a common algorithm implemented in firewalls and NATs that helps devices be aware of information available at the application layer that is within the IP payload data Being aware of the application layer the ALG inspects packets for embedded IP address and port information and can perform NAT without breaking applications such as FTP RTSP ICMP SIP H323 SKINNY and MGCP Compared with regular NAT ALG allows a pinhole to be dynamically created to permit the data exchange for the session and correlates control and data sessions ALG can use the same common timeout value or they can keep them independent Firewalls policies can be configured to trigger the ALG module based on either application or by service type ALGs solution for NAT traversal is similar with the solution based on SBCs explained in the next section Session Bo
50. number of simultaneously active calls 2 Set the parameters of Voice Session Properties for quality of voice PESQ analysis PESQ computes the quality of voice scores by comparing the degraded clip the audio received with the reference The clip used as the reference on the receiving side should match the clip played on the other side a Inthe Talk Parameters tab select the Overwrite playback activity settings check box b Inthe Talk Parameters tab set the clip in Payback Settings to a voice clip from the Speech Clips pool PN 915 2611 01 Rev H June 2014 90 Test Case Subjective Quality of Voice Yoice Session Properties Miel Talk Parameters f Listen Parameters Advanced Settings Output Settings Delay Before Execution Static Expression jo m w Oyermrite playback activity settings PCM S000 Hz bkk ainiai Volume 20 dEr Please select a clip item from the Qos Pool or from the Wave Pool i Play i tii Speech Clips Simple Prompts Repeat continuou Use TalkTime for 0 Female ale_ Mixa 1 FemaleMale Mine H2 Female ale_ Mix 3 Males Seqe H4 Malez_Segl R a Malel_Seq4 Plays one or more WAVE F 6 Malel_Seqs hd wave Files can have the Fe Coding PCM 4 LAaw I Cancel Restore Defaults OK Eenez f Use Global setting Sampling Frequency 6 Figure 76 Setting the voice clip to be played c Inthe Talk Parameters tab select
51. om Figure 66 The expected characteristic of throughput PN 915 2611 01 Rev H June 2014 69 Test Case Determining the Maximum Number of Concurrent Calls 2 What is the call rate Table 16 Call rate statistics Statistic Name Questions Have the calls been attempted continuously at a Calls Attempted per Second constant call rate during the Sustain Time Calls Connected per Second How do the calls attempted rate and the calls pee connected rate compare with each other 3 Have any call failures been reported Check the Calls view Table 17 Call Statistics Statistic Name VELT Questions Calls Attempted 1 Has any call attempt failed Calls Connected Compare Calls Received c Calls Attempted and Calls Received Calls Answered d Calls Attempted and Calls Connected End Calls Initiated 1 Has any attempt to hang up the call failed End Calls Received Compare c End Calls Initiated and End Calls Received d Calls Attempted and End Calls Received Have all the calls attempted ended successfully Compare End Calls Completed 2 Calls Connected and End Calls Completed Why do we need to compare End Calls Completed with twice the number of calls connected 4 Have any scenario loop failures been reported Check the Loops statistics view Table 18 Statistics highlighting the pass fail result based on the call flow execution Statistic Name Value Questions Total Loops a 1
52. reason may be the miss configuration of some of the spans if many spans are used the mismatch of source and destination dial plan lack of resources on DUT to route all the calls It is possible on some DUTs to observe the number the number of Attempted Calls equal with the number of Connected calls but different then the number of Received Answered Calls In this case the DUT completes the SIP call legs but fails to set the PSTN calls showing a functionality error E calls PSTNDigital ERE Attempted Calls Connected Calls Failed Attempted Calls Atkempted Calls E connected Calls Failed Atbempted Calls PN 915 2611 01 Rev H June 2014 170 Test Case Determining the Capacity of a VoIP to PSTN Gateway 2 Have any call failures been reported Check the Calls view Table 59 Call failure statistics Statistic Name Questions SIP Failed Attempted Calls 1 Is any called fail or rejected PSTN Rejected Calls Note If there are calls failures on the SIP side the SIP Message stats may be checked to find the reason of failures 5xx responses means the internal DUT error 401 or 407 responses means the DUT has the authentication enabled and the call flow should use also authentication 404 response means the destination phone number is not configured on the DUT The SIP event viewer shows the errors in the SIP call flow 3 Have any scenario loop failures been reported Check the Loops statistics view Table 60 S
53. regular PSTN networks The performance impact of common complex features such as Class 5 telephony features for example call transfer call conference call parking call waiting is underestimated In practice the complexity and message overhead added by those features may lead to system stability issues if minimal testing was performed at low scale using just a few phones Security A VoIP system is susceptible to a number of attacks and threats including denial of service DoS attacks eavesdropping registration and session hijacking and server impersonation Multiple levels of security are implemented at the transport and application levels to prevent attacks Protocols such as IP security IPsec TLS and SRTP add additional complexity and system performance can decrease PN 915 2611 01 Rev H June 2014 5 Test Case Determining the Maximum Call Setup Rate CPS Test Case Determining the Maximum Call Setup Rate CPS It is important to determine the maximum call rate for devices that handle VoIP call setup and teardown phases A few examples of such devices are Soft switches IP PBXs Call managers session border controllers VoIP application layer gateways Voice mail and interactive voice response servers IVRs Application servers Conference bridges The primary performance metric assesses whether the attempted call rate is sustainable for a long time without call failures or affecting users experience The
54. test objective been achieved check the Loops Rate view The value of the Loops per Second statistic should follow the time line specified in the test objective In this example the expectation was for a steady value of 500 loops per second If the graph shows variations in the Loops per Second value that were not specified in the advanced time line then the test has been misconfigured or the DUT cannot maintain the desired rate The IxLoad configuration should use enough emulated endpoints to maintain a specific rate If the registration of one endpoint takes 2 seconds then 1000 endpoints are required to maintain a registration rate of 500 per second In this example 8 000 endpoints are defined To maintain a rate of 500 loops per second each loop may take up to 16 seconds 8 000 500 If one loop requires more than 16 seconds then there PN 915 2611 01 Rev H June 2014 114 Test Case Determining the Maximum Transaction Rate for VoIP Protocols won t be sufficient endpoints available to start a new loop after 16 seconds causing the loop rate to drop E Loops per Second ile eu 44 POO dile ii 1 485 S04 20 rai ae POS litt 140 1 56 fle Figure 104 Loop rate view 2 To determine if the registration rate has been achieved check the Registration Rate view Statistic Name Questions i i a ess s s s sSS Has the Registration Initiated rate attained Registration initiated a constant rate during the Sustain Time
55. the Overwrite playback activity settings check box PN 915 2611 01 Rev H June 2014 Test Case Subjective Quality of Voice d Inthe Listen Parameter tab select the Perform QoV measurements check box and select the same clip as in the previous step Leave the Listen duration at its default value oice Session Properties Mel X i Talk Parameters Listen Parameters Advanced Settings Output Settings s Oyeme playback activity settings kien oetIrigs i Listen duration 20033 m E f Use Talk Time for all objectives except Channels Perform Qoy measurements Clip FemaleMale_Mix3 Output level 20 dBm Output Yah irna 20 AR AI N FFARR EON obra Yoice Clips Items Please select voice clip from the list 0 FemaleMale_ Mia Femalel ale Mix Malez_5egq2 Malet Seql Male Seq4 Malel_Seqs Malel Seqz Malel Seq D Oo e oo Po Plays one or m wave Files can Coding PCh Sampling Fre Restore Defa Figure 77 Set the reference for PESQ analysis on the listener side PN 915 2611 01 Rev H June 2014 92 Test Case Subjective Quality of Voice 3 At the activity level on the Audio settings tab select the Perform QoV check box Set the parameters in accordance with the volume of traffic desired and available resources One Ixia VQM load module can perform real time PESQ analysis on 300 audio streams The test described here uses a single VQM load module and because RTP tr
56. the call duration multiple Voice Session Call duration 20 sec functions should be placed in the test scenario TIP a loop controlled by the Variable Test and Variable Set functions can be used the language and M ale English Male speaker s gender 25dBm 30dBm 35dBm Variation of the Active level parameter may influence the Quality of Voice if the DUT implements AGC automatic gain control or VAD voice activity detection Use simplex RTP traffic by replacing the Voice Session function on one channel with the Talk function and the Voice Session function on the other channel with Listen function The parameters of these functions should be changed to play a speech clip and to perform QoV measurement respectively RTP traffic direction Full duplex Use half duplex RTP traffic by replacing the Voice Session functions on both channels with a sequence of Talk Listen and Listen Talk functions respectively The parameters of these functions should be changed to play a speech clip for the Talk functions and to perform QoV measurement for the Listen functions respectively DUT Test Variables Table 39 DUT test variables PESQ test Parameter Name Current Value Additional Options Codec on public network G 729A G 729B G 723 G 726 PN 915 2611 01 Rev H June 2014 96 Test Case Subjective Quality of Voice Results Analysis The results analysis is done by comparing PESQ scores with expected values If th
57. to Measure NAT PAT Performance This example is illustrated below mer ae age eee eae Si poea e arnal Makaa Bie First connection takes the Internal Network s first available IP address from the pool p private NAT NAPT 92 168 1 100 24 i 1 N Second S ia Connection ON 192 168 1 102 24 CE 75 83 202 16 22 QR Router E P a Third request takes the 3r available address from the pool o Y 75 83 202 14 22 A The second 75 83 202 15 22 A aa takes he 2 4 available address connard 192 168 1 1 24 Figure 113 Dynamic NAT example Dynamic NAT is more common in implementation that combines NAT and NAPT rather than using it as a standalone mode In many cases basic NAT allows hosts in a private network to transparently access the external network and enables access to selective local hosts from the outside Organizations with a network setup predominantly for internal use with a need for occasional external access are good candidates for this scheme This does not resolve the main problem however which NAT needs to resolve I Pv4 address depletion Many organizations have multiple network nodes running TCP UDP applications which require Internet access but only one public IP address assigned to their remote router The NAPT mode resolves this issue by permitting multiple nodes in a local network to simultaneously access remote networks using the single IP
58. to tear down a call e Message retransmissions counts the number of retransmissions required to deliver control plane messages across the network A high number of retransmissions received or transmitted by a DUT can drop the overall performance of the DUT and cause stability issues The number of messages retransmitted must be maintained below 5 percent in a deployed IP telephony system and should be as close as possible to zero when testing a DUT in isolation e Call establishment ratio measures the ability of a called party to successfully connect a call and establish a conversation Call Pctablickment Ratio of Calls Connected 100 96 all Establishment Ratio of Calls Attempted 1 e Call completion ratio measures the ability of a called party to successfully connect the call and to successfully complete the call by initiating or receiving the appropriate disconnect request of End Calls completed 100 LS of Calls Attempted Call Completion Ratio When the DUT simultaneously handles signaling and media it is important that media traffic is enabled for the calls Media traffic must be tested to ensure that it meets the QoS requirements The following standard metrics can be used to help assess the QoS for RTP based voice traffic e Assessing media transport o Packet loss o Jitter o One way delay o RTP packets sent versus RTP packets received o Inbound and outbound throughput Maximum conse
59. traffic characteristics and the desired QoS The video codecs compensate well with delays up to 200ms from end to end but higher values can trigger changes in bitrate or resolution However a 200ms delay has a significant effect on audio quality Typically vendors implement methods of informing the user about large latencies and the possible effect on the call quality If end to end delays larger than 400ms are recorded the call might not be sustained and call termination might be encountered However in certain situations like intercontinental calls or satellite teleconferencing these limits can be increased to allow a sustainable call This implies a lower rating in user experience index due to the amount of time required for a spoken message to be acknowledged For these situations even if technologically the telepresence application will be feasible the user s expectations should be set accordingly Besides the effect of delay delay variation can degrade the quality of the received media To preserve high quality video the telepresence maximum jitter should be within 10 of the maximum delay accepted For example if the end to end delay accepted before quality change is 200ms the maximum tolerated jitter should not exceed 20ms Depending on the coding and decoding techniques different vendors can adjust their thresholds to produce a better user experience and a higher quality image Packet loss is the network impairment that most affect
60. traffic is insignificant but may be an issue if large numbers of devices register at the same time The primary causes for registration flooding are e Power outages when power returns to a region all at once all VoIP devices will initiate registration transactions at the same time e Defective infrastructure device if an edge device goes down serviced endpoints will attempt to re register causing signaling messages to flood the backup device PN 915 2611 01 Rev H June 2014 105 Test Case Determining the Maximum Transaction Rate for VoIP Protocols These situations are relatively rare and are addressed in the protocol specifications for example Retransmissions and Configurable timeouts The system will converge and all devices will eventually become active The most significant source of high volume signaling traffic is because of the periodic transmission of messages related to the following e Keep alive mechanisms e Content embedded with signaling messages as in Instant Messaging e Misconfigured or defective endpoints e Network attacks SIP itself does not define a keep alive mechanism As a result there are various implementations that re use other SIP messages Common mechanisms include use of the Register message the Invite message during a call or the Option message although some devices do not support this message The remainder of this booklet discusses the methodology and configuration steps needed to det
61. will look as shown below SRCIP S sSs SRC PORT DEST IP DEST PORT 172 16 100 2 172 16 100 9 200 OK SIP 2 0 Via SIP 2 0 UDP 172 16 100 9 5060 From lt sip 8184443118 myserviceprovider com gt tag 5000c5000 To Alice lt s1ip 8184443118 myserviceprovider com gt tag 6000c6000 Contact Alice lt sip 8184443118 172 16 100 9 5060 gt Content Length 0 Expires 1800 Figure 134 Structure of the message 4 200 OK received by SBC As previously mentioned when the SBC forwards the 200 OK response back to the IP phone it sends the response to the public address of the CE NAT gateway 75 83 202 16 1025 It uses PN 915 2611 01 Rev H June 2014 140 Test Case Using VoIP to Measure NAT PAT Performance the Call ID and transaction information received in the initial REGISTER request The response will use the private IP address of the IP phone The structure of the 200 OK response is shown below SRCIP SRC PORT DESTIP DEST PORT 193 16 148 244 75 83 202 16 1025 200 OK SIP 2 0 Via SIP 2 0 UDP 193 16 148 244 5060 From lt sip 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt 51p 8184443118 myserviceprovider com gt Contact Alice lt sip 8184443118 192 168 1 100 5060 gt Content Length 0 Bxpires 180 Figure 135 Structure of message 5 200 OK On receipt of the response the CE NAT gateway looks up the destination address in its translation table and applies the re
62. with Media Clips library This contains all the multimedia files required by the telepresence scenario emulation Any custom files can be loaded in the Media Library pool that can represent real recordings of events or actions Step by Step Instructions The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which can be used to change the behavior of the test 1 Start IxLoad 2 Select New from the File menu options and then select Templates Reports Views e Save AS Available Resources Save gt 1 fGhHome VoIPSip S Open Repository J ka Export Empty Contig KLA Import E Items Figure 209 Path to Templates configuration files PN 915 2611 01 Rev H June 2014 224 Test Case Measuring Quality of Experience for Multimedia VoIP Calls 3 Inthe new page go to VoIPSip folder Empty Config Items OF VoIPMGCP Figure 210 VoIPSip configuration templates 4 Select the TelepresenceSample rxf file from the Telepresence folder The configuration is loaded when the files is accessed lt gt 1 Home gt VoIPSip gt Telepresence Figure 211 IxLoad telepresence configuration template 5 Access the originating NetTraffics to change the network connectivity details by accessing the Network1 IP connectivity details Networks and Traffic New Traffic Flow om A gt bg Sta
63. with each other RTP MOS Worst l How does the RTP MOS Worst score compare with the max theoretical score for the CODEC used Are there any times without an instantaneous MOS RTP MOS Instant l value Best Avg Worst How frequent are the changes in the instantaneous MOS values RTP MOS Per Call l How are the MOS per Call statistics compared with the Best Avg Worst RTP MOS Best and RTP MOS Worst statistics PN 915 2611 01 Rev H June 2014 71 Test Case Determining the Maximum Number of Concurrent Calls Table 21 Basic RTP QoS Statistics see RTP QoS and RTP Advanced QoS statistics views Statistic Name Value s Questions Are there any differences between RTP Packets RTP Packets Sent Sent and RTP Packets Received RTP Packets Received Do the differences match the value of RTP Lost RTP Packets Lost Packets RTP Delay Variation Jitter us RTP Interarrival Jitter us What is the max Delay Variation Jitter What is the max Interarrival Jitter RTP One Way Delay us Is the One Way Delay higher than 100 ms Table 22 RTP Jitter Distribution statistics Statistic Name Name VEUT E Questions mreze Jitter up to 1 ms Jitter up to 3 ms Jitter up to 5 ms Jitter up to 20 ms Jitter up to 40 ms Jitter over 40 ms Packets with Delay Variation 1 Assuming Jitter was reported what is the Jitter up to 10 ms distribution of the Delay Variation Jitter value
64. with the port number used by the SIP Server typically 5060 IxLoad will send the messages to the Server Address on port Server Port e Set the Domain Name or Local IP option to match the one set on the server for example voice ixiacom com PN 915 2611 01 Rev H June 2014 2 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Trafficl Make_Call VoIPSip Peer Scenario Execution ial Plan SIP automatic 715 Cloud Codecs RTP Aud Video Fax 38 Fax 7 30 SRT Enable signaling on this activity SIP Port 5060 if unchecked all SIP script Functions will be SKIPPED Transport settings o Use external server Maximum message size on UDP 1024 _ Override transport specified in scenario TCP send immediate Authentication UAC Username Anonymous Construction of SIP messages Password AKA authentication settings _ Override default contact settings Select configuration lt None gt _ Override default destination domain name or host por _ Use Tel URI scheme for Source Type Of Service _ Use Tel URI scheme for Destination TOS DSCP Transfer address verify all settings Restore defaults Transport settings Use external server Maximum message size on UDP 1024 Server address 61 15 1 1 m Override transport specified in scenario Server port 5060 i Domain name or local IP voice ixiacom com _ TCP send immediate _ Enable FQDN resolution
65. 0 24 and 192 168 1 200 24 They are configured to use as a gateway the IP address of the CE router on the private interface 192 168 1 1 In this example the CE router uses the public IP address 75 83 202 16 22 to connect to the public network 192 168 1 100 24 Internal Network i e e private NAT NAPT Network public l I 192 168 1 1 24 ee ee Figure 107 NAT concept explained As you can see all traffic from the hosts is sent to the public network through the CE router which performs NAT on these packets The packets are then forwarded to their destination via the provider edge PE router which is configured as a gateway for the public interface of the CE router Now let us assume that host 192 168 1 100 24 uses an HTTP client to retrieve an HTML page from an external server with the address 209 132 176 30 www redhat com The client initiates a GET request using the source IP 192 168 1 100 and source port number 32 000 with the destination 209 132 176 30 and destination port 80 PN 915 2611 01 Rev H June 2014 121 Test Case Using VoIP to Measure NAT PAT Performance Modification of the IP Header for the OUTGOING packet request The IP Header of the received packet 192 168 1 100 209 132 176 30 Interface Interface Translation 192 168 1 100 lt gt 75 83 202 16 N The modified IP Header sent on the public interface Pb 75 83 202 16 32 000 209 132 176 30 ow Figure 108 Processing of the first
66. 1 01 Rev H June 2014 Test Case VoIP Quality of Service in Converged Networks Test Variables Depending on the characteristics of the DUT the test should be repeated with following variations Table 35 Test tool variables Parameter Name Current Additional Options VEUT IP Type aE Static IPv6 IPv4 or IPv6 DHCP In addition to HTTP traffic you may add other types of activity ftp Data Traffic http smtp pop3 to better simulate the type of traffic generated consumed by subscribers and passing the DUT Video traffic in this example is unicast video on demand Multicast idee TANE VoD IPTV can be added or replace the VoD traffic Audio Codec G 711 Lower bit rate codecs G 729 G 723 request lower bandwidth DUT uLaw may scale better for a bit rate codecs 5 ms 10 ms 30 ms RTP overhead is significant for small packets and the number of RTP packets increases with a reduction in packet time Audio Codec At 20 ms if there are 10 000 packets sec for 100 full duplex calls the Packet Time number of packets becomes 40 000 sec at 5 ms packet time IP Ethernet throughput is doubled by changing the packet time from 20 ms to 5 ms Ua os CP ior Best Class 1 2 3 4 Express Forwarding Control or Custom according to ignaling or RTP Eff traffic ort the settings on the DUT Total Inbound 300 Increase the volume of traffic to the maximum level supported by the Throughput Mbps DUT PN 915 2611 01 Rev H June 2014 83 Test Case VoIP Qu
67. 11 01 Rev H June 2014 187 Test Case Measuring Quality of Experience for Voice Calls in LTE Test Case Measuring Quality of Experience for Voice Calls in LTE Overview With the migration of mobile networks to all IP networks defined by the LTE specification there is a need to migrate the voice and SMS services as well Today there are several options for carrying voice over LTE using different technologies e CSFB Circuit Switched Fall Back The circuit switched fallback CSFB option for providing voice over LTE has been standardized under 3GPP specification 23 272 Essentially LTE CSFB uses a variety of processes and network elements to enable the circuit to fall back to the 2G or 3G connection GSM UMTS CDMA2000 1x before a circuit switched call is initiated The specification also allows for SMS to be carried as this is essential for very many set up procedures for cellular telecommunications To achieve this the handset uses an interface known as SGs which allows messages to be sent over an LTE channel e SV LTE simultaneous voice LTE SV LTE allows running packet switched LTE services simultaneously with a circuit switched voice service SV LTE facility provides the facilities of CSFB at the same time as running a packet switched data service This is an option that many operators will opt for However it has the disadvantage that it requires two radios to run at the same time within the handset This has a serious impact on bat
68. 16 100 2 5060 1 INVITE SDP DST 193 16 148 244 5060 E Alice l 192 168 1 100 62 i 2 INVITE SDP Session Border amp Enterprise NAT NAPT Controller Service Provider s WAN es nN Wii Core Network 192 168 1 1 UW 193 16 148 244 4 200 OK SDP 6 200 OK SDP 75 83 202 16 5060 wsRO 172 16 100 2 5060 SRC 192 168 1 1 5060 1025 5 200 OK SDP DST 172 18 0 0 SAGL DST 192 168 1 100 5060 SRC 193 16 148 244 5060 Media Gateway DST 75 83 202 16 1025 172 16 100 3 INVITE SDP INVITE SDP c IN IP4 192 168 1 100 c IN IP4 192 168 1 100 m audio 10000 m audio 10000 INVITE SDP c IN IP4 172 16 100 10 m audio 25000 200 OK SDP 200 OK SDP 200 OK SDP c IN IP4 172 16 100 3 c IN IP4 193 16 148 245 c IN IP4 193 16 148 245 m audio 30000 m audio 20000 m audio 20000 192 168 1 100 75 83 202 16 0 172 16 100 3 10000 15000 30000 Figure 137 How an SBC resolves the media traversal problem PN 915 2611 01 Rev H June 2014 142 Test Case Using VoIP to Measure NAT PAT Performance Remember that Alice s phone must be configured to use the public IP address of the SBE SBC 193 16 148 244 5060 as its outbound proxy In this way Alice s phone will send all the requests via the public interface of the SBC When the 1 INVITE request is sent by Alice the SBC detects the presence of a NAT device between them by comparing the source IP address and port number of the IP header with the IP address and port number included into t
69. 2 10 205 17 114 1 12 10 205 17 114 1 12 Event Tite Loop 290694 SIP Parser Error in Unrecognized Header Line Text lt dm person gt Loop 290694 SIP Parser Error in Unrecognized Header Line Text lt presence gt Loop 290696 SIP Parser Error in Expires Header Text Expires 999999999999 Loop 290697 SIP Parser Error in Expires Header Text Expires 9a9b9c9d9e9T Events In View 20000 Figure 238 Event Viewer logged messages sample 8 Has the test reported any errors Once the DUT detects the TDoS attempt or the system is at the maximum processing limit various errors might occur Check the information displayed in Errors view PN 915 2611 01 Rev H June 2014 255 Test Case Telephony Denial of Service Table 91 Statistics used to determine the SIP Registration rate Statistic Name Questions Are there any timeout errors or transport errors reported Are there any Call Flow errors reported Could you identify the cause Are there any increasing values for the Variable Extraction Statistics reported Verify if the proper variables are used to SIP Extract Variables construct the test procedures errors SIP Internal errors a Conclusions This test methodology provided details of how to emulate with IxLoad a Voice SIP endpoint initiating forged SIP Register messages towards the DUT in the attempt of a Telephony DoS with malformed messages The scenario
70. 2 Select the Forcefully Take Ownership check box 3 Select the Reboot Ports before Configuring check box 4 Set CSV Polling Interval to 1 second Navigation 4 Test Options Test Configuration Test Run Statistics Overview v Reboot Ports Before Configuring Throughput Stat Units Kbps Networks and Traffic E A New Traffic Flow 2 1 Originate E Trafficl Network Network eee _ Enable Conditional view DUT 3 1 Terminate Network Failure Threshold 0 al So _ Release Configuration After Test _ Enable TCP Advanced Stats _ Enable Frame Size Distribution Stats oy Traffic2 Network Timeline and Objective Port Assignments Hardware Test Options Card Family ASM XMV12 Options _ Enable Network Diagnostics _ Allow multiple 1G aggregated groups of ports Figure 40 Configuration example for Test Options Configuring Port Assignments 1 From the Test Configuration panel click Port Assignments 2 Add your chassis 3 Assign the port s for Network1 hosting the Make_Call activity PN 915 2611 01 Rev H June 2014 38 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 4 Assign the port s for Network1 hosting the Receive_Call activity Navigation 2 Port Assignments Test Configuration Test ra Overview ge E Networks and Traffic E 8 New Traffic Flow B IT Originate oe Traffic l Network Eifi Terminate ra Tel nee at Statistics Ru
71. 4 4 Table 37 R Factor R Factor User Satisfied 61 70 Many users dissatisfied 51 60 Nearly all users dissatisfied 0 50 Not recommended R Factor and PESQ both characterize a voice transmission system the former considers the packet networks impairments while the latter compares the received audio signal with the expected signal Beside the effect of the impairments of the transmission network PESQ also captures the effects of trans coding voice activity detector echo cancelation and any other type of audio signal alteration Because the PESQ algorithm is computationally intensive it is not practical to use it for testing the speech quality on high scale devices or systems However if E Model is used exclusively some issues may remain hidden if the system performs audio signal processing The best practice is to combine the two methods and perform E Model measurement on all calls and PESQ on a smaller percentage of them PN 915 2611 01 Rev H June 2014 88 Test Case Subjective Quality of Voice Objective The test that is described here will determine if the DUT provides the desired level of load without quality of voice degradation If degradation is observed the cause will be determined Setup IXIA Simulated SIP User Agents IXIA Simulated SIP User Agents Application Layer Gateway Figure 74 Typical topology for QoV measurement in an end to end voice communication system The DUT used in this example is
72. 4000 Iteration Time 0000 07 50 3000 Iterations 2000 Time bo First Iteration 0000 00 00 1000 Iterations 1 Time Between Iterations A 0 00 00 0 00 45 0 01 30 0 02 15 0 03 00 0 03 45 00430 0 05 15 0 06 00 0 06 45 0 07 50 Figure 61 Configuration example for the H323 concurrent calls test objective PN 915 2611 01 Rev H June 2014 65 Test Case Determining the Maximum Number of Concurrent Calls Configuring Test Options 1 From the Test Configuration panel click Test Options 2 Select the Forcefully Take Ownership check box 3 Select the Reboot Ports Before Configuring check box 4 Set CSV Polling Interval to 7 second Navigation 7 Test Options Test Configuration Test Run Statistics Testi Forcefully Take Ownership CSV Polling Interval 1 Senne Overview Reboot Ports Before Configuring Throughput Stat Units Kbps ic a i C Release Configuration After Test _ Enable TCP Advanced Stats New Traffic Flow J1 Origi _ Enable Frame Size Distribution Stats ginate Network E Trafficl Network i DUT _ Enable Conditional View Ei Terminate Network Failure Threshold 0 o Sa Traffic2 Network Timeline and Objective Port Assignments Hardware Test Options Card Family 45M XMV12X Options m Enable Network Diagnostics Allow multiple 1G aggregated groups of ports Figure 62 Configuring Test Options Configuring Port Assignments 1 From the Test Configuration panel c
73. 75 83 202 16 SRC IP 192 168 1 102 75 83 202 16 22 SRC PORT 1025 SRC PORT 5060 DST IP 209 132 176 300 E DST IP 209 132 176 30 DST PORT 3478 192 168 1 1 24 E l EN ga Q Figure 126 STUN client initiating a Binding Request e The STUN client initiates a STUN binding request using the source address 192 168 1 102 and port number 5060 the packet is sent to the external STUN server at 209 132 176 30 which listens on the standard STUN port 3478 e The NAT gateway receives the STUN binding request translates the 192 168 1 102 address to the public IP address 75 83 202 16 and the private port 5060 to the public port 1025 e On receipt of the request the STUN server issues a STUN binding response from 209 132 176 30 3478 with the destination set to the public IP address port used by the NAT gateway to initiate the STUN binding request 75 83 202 16 1025 The payload of the STUN binding response message includes the MAPPED ADDRESS field set to the public IP port address of the NAT gateway 75 83 202 16 PN 915 2611 01 Rev H June 2014 135 Test Case Using VoIP to Measure NAT PAT Performance STUN client 192 168 1 102 5060 receives the STUN Binding Response from the public STUN server 209 132 176 30 3478 ee ee Be ee ee ee wl Private Network Internal j Public Network External I NAT NAPT 192 168 1 102 24 l SRCIP 209 132 176 30 75 83 202 16 22 SRC POR
74. AT PAT Performance Static NAT can also be used for outbound sessions initiated from the private network While this is not acommon use it can still be useful in certain situations As described in the NAT Concept section basic NAT requires a public IP address for each concurrent connection that may pass through the NAT The bindings between specific private IP addresses and specific public addresses can be statically defined ee oe a P a amp 75 83 202 16 22 75 83 202 15 22 e oo k Figure 112 Two concurrent NAT translations Dynamic NAT Static NAT provides a one to one private to public static IP mapping whereas dynamic NAT provides the same functionality by using a dynamic mapping to the public IP addresses based on a group of publicly available IPs Let s assume a simple configuration consisting in three hosts located in the private network and a NAT router that is configured with a dynamic pool of three public IP addresses When a new outgoing connection is received from the private network the NAT router substitutes the private IP address with the first available address in the pool When the second host initiates an outgoing connection assuming that the first host has its connection still active the NAT router will use in the secondary public IP address for the translation Similarly the 3 host will receive the 3 address from the pool PN 915 2611 01 Rev H June 2014 125 Test Case Using VoIP
75. Are the Successful Loops and Total Loops equal 2 Have any failed loops aborted loops or warning eee Loops ___ loops been reported Failed Loops Z o Note failed aborted and warning loops highlight Aborted Loops k failures at the scenario level Warning Loops ee PN 915 2611 01 Rev H June 2014 70 Test Case Determining the Maximum Number of Concurrent Calls 5 Has the QoS for signaling met the expected quality Check the Call Times and Delays Statistic views Use the maximum value reported Table 19 Statistics used to determine the QoS for the SIP signaling Statistic Name Value Questions max avg min Is the maximum Call Setup Time less than 4 Call Setup Time seconds End Call Time a Is the maximum End Call Time less than 2 Talk Time seconds Media Delay TX RX rr Is the maximum Media Delay Tx or Rx less than 4 seconds Post Dial Dela a Is the maximum Post Dial Delay less than 2 seconds Is the maximum Post Pickup Delay less than 2 Post Pickup Delay seconds For every stat listed in this table compare its value distribution in time 6 Has the QoS for media met the expected quality Check the RTP MOS RTP QoS RTP Advanced QoS RTP Jitter Distribution RTP Consecutive Lost Datagram Distribution and RTP Streams statistic views Table 20 MOS Statistics Statistic Name VETRE Questions How do the last values reported by the RTP MOS Best RTP MOS Best and RTP MOS Worst compare
76. Black Book Ixia Edition 10 Voice over IP http www ixiacom com blackbook June 2014 Voice over IP Your feedback is welcome Our goal in the preparation of this Black Book was to create high value high quality content Your feedback is an important ingredient that will help guide our future books If you have any comments regarding how we can improve the quality of this book or suggestions for topics to include in future Black Books please contact us at ProductMUgmtBooklets ixiacom com Your feedback is greatly appreciated Copyright 2014 Ixia All rights reserved This publication may not be copied in whole or in part without Ixia s consent RESTRICTED RIGHTS LEGEND Use duplication or disclosure by the U S Government is subject to the restrictions set forth in subparagraph c 1 ii of the Rights in Technical Data and Computer Software clause at DFARS 252 227 7013 and FAR 52 227 19 Ixia the Ixia logo and all Ixia brand names and product names in this document are either trademarks or registered trademarks of Ixia in the United States and or other countries All other trademarks belong to their respective owners The information herein is furnished for informational use only is subject to change by Ixia without notice and should not be construed as a commitment by Ixia Ixia assumes no responsibility or liability for any errors or inaccuracies contained in this publication PN 915 2611 01 Rev H June 2014 lil
77. ContentLength O SIP_Route gO SIP_RecordRoute SIP_Authorization O SIP_ProxyAuthonzation O SIP_Www Authenticate O SIP_ReferTo gO SIP_Replaces O SIP_RetemedBy O SIP_Event O SIP_Subscription State Send a sip request message Restore Defaults June 2014 Enabling automatic variables processing 249 Test Case Telephony Denial of Service 14 Repeat steps from 4 to 10 to add a new procedure to the call flow and enable the offer of SDP information in the originating requests In the Request Properties configuration tab enable the checkbox of the Send audio SDP parameter Set the option to Offer from the available drop down options The DUT should ignore these types of requests with SDP information and no stability issues should be noticed i ap Send Request P operties Params Behavior Flow Manager Extract Variables Authentication Output _4 I REGISTER AUTO REQUEST URI SIP 2 0 Via AUTO VIA From AUTO FROM To AUTO TO Call ID AUTO CALL ID Cseq AUTO CSEQ Contact AUTO CONTACT Max Forwards 70 Content Length AUTO CONTENT LENGTH Expires 36500 Create From Template Load From File Created from template Iw Case sensitive I elay before execution Static Expression Message body Send custom message body Edit Options Change case Edit Custom Send a sip request message Restore Defaults F
78. DEC type must be considered The CODEC that is used is even more important when the DUT acts as a transcoder for example converts the voice from G 711 to G 729 PN 915 2611 01 Rev H June 2014 10 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Overview The goal of this test methodology is to determine the maximum peak load measured in calls per second supported by a DUT that implements at least one of the following SIP logical components defined by the SIP protocol e SIP proxy server e SIP redirect server e SIP back to back user agent The primary test metric is the number of calls that the DUT sustains for a long time while providing the desired user experience In addition the QoS metrics discussed in the Max Call Setup Rate section are required to determine pass fail criteria Those pass fail criteria may be different for a DUT that is isolated by the test tool as opposed to a complete IP telephony system that is passing calls over a WAN For example the post dial delay should not exceed 250 ms if the test tool isolates the DUT while a post dial delay of 2 seconds is acceptable when the DUT is a complete IP telephony system where the post dial delay is measured across the WAN Obviously the QoS constraints are more restrictive when running with the DUT in isolation because practical deployments include a chain of DUT
79. FODN resolution Authentication UAT Username Anonymous Construction of SIP messages Password 4K authentication settings Override default contact settings Edit Contact Select configuration lt None gt Override default destination domain name or host por Edit configurations een E E ree Use Tel URI scheme For Source __ Type OF Service _ Use Tel URI scheme For Destination Tosipscp Best Effort 0x00 Transfer address verify all settings Restore defaults Figure 33 SIP Settings for back to back configuration Transport settings Use external server Maximum message size on UDP 91024 Server address 61 15 11 _ Override transport specified in scenario Server port cs gg UCF Gnl Domain name or local IP vOICe ixiacom com _ TCF send immediate _ Enable FQDN resolution Registrar server Figure 34 Sample configuration with a SIP Server 61 15 1 1 acting as an Outbound Proxy PN 915 2611 01 Rev H June 2014 33 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring Automatic Behavior for the Receive_Call Activity 1 Click the Automatic tab 2 Select the Enable Retransmission Ignore received retransmissions and Retransmit ACK check boxes and leave the default values for the T1 and T2 timers a T1 500 ms default b T2 4000 ms default Enable session timers Cx PIPacion w i alue per type of message Timeout and retransmisions
80. Figure 157 Associate a span to a PSTN Network Range Configuring PSTN Board Clock Settings For T1 E1 boards configuring the clock settings is important With respect to the ECTF H 110 specification the telephony boards supported by IxLoad can be configured to use group A clock lines only fallback to group B is not supported configured as either standalone that is decoupled from the CT bus master or slave For the current release the clock settings apply at the ADM carrier board level only it is not possible to synchronize the clocks of telephony modules from different adaptor modules PN 915 2611 01 Rev H June 2014 167 Test Case Determining the Capacity of a VoIP to PSTN Gateway To configure PSTN board clock settings do the following 1 In the Assigned Ports or in the PSTN Interfaces Assignment window right click an assigned ADM board and click Clock Settings In the Clock Settings window that appears all PSTN boards installed in the ADM carrier board are displayed ixt Clock Settings Dialog gt Boards on a port can be configured as clack master or standalone A single board can be assigned as source For Master Clock within a port if the clachl lt is to be derived From network then a single physical span of this board can be specified as source otherwise it will be derived From board internal oscillator PSTN Local Clock Settings ees 1 TOB E1 T1 08 Standalone Internal ha Mot driven ha 10 205 19 87 ae TOB E1
81. Fort 1 4 12 IxLoad 52 554 Configured For 1 on Card 5 4CCELERON NP bei Port 1 5 2 IxLoad 52 554H Configured for 1 E Fort 1 5 3 Ixboad S2 S54H Configured for 1 E Port 1 5 4 xLoad S2 554H Configured for 1 ah Statistics E Fort 1 5 5 Ixload S2 S54H Configured for 1 show IP Assignments 3ssion PSTN Interface Figure 63 Assigning test ports Running the Test 1 Click Run to start test execution 2 IxLoad will automatically display Statistic Views after the execution starts Configuration Highlights for H 323 Fast Start Mode H 323 version 2 introduces a new method of call setup called fast start or fast connect In this mode an H 323 endpoint will bypass some steps to make call setup faster In addition to the speed improvement fast start allows the media channels to be operational before the CONNECT message is sent which is a requirement for certain billing procedures To change the configuration to use H 323 fast start mode repeat the following steps for both H 323 activities 1 Click the H 323 configuration page of Make_Call 2 Verify that the Enable signaling on this activity check box is selected PN 915 2611 01 Rev H June 2014 67 Test Case Determining the Maximum Number of Concurrent Calls 3 Select the Enable FastStart check box Traffici Make Call VoIPH323 Peer Enable signaling on this activity iF unchecked all H323 script Functions will be SKIPPED H323 Specific
82. H June 2014 44 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Conclusions This configuration covered the main parameters of the SIP Peer activity using a practical example allowing the user to control the call rate while running performance tests The results section covered the main statistics that may highlight an issue PN 915 2611 01 Rev H June 2014 45 Test Case Determining the Maximum Number of Concurrent Calls Test Case Determining the Maximum Number of Concurrent Calls Determining the maximum number of active calls with media is necessary for devices and networks that handle media traffic wnere media may include text audio video or fax A few examples of such devices are session border controllers Application layer gateways Media gateways Voice mail servers Interactive voice response systems Media transponders Media proxy servers The primary performance metric is the number of active calls that can be maintained by the DUT for a long period without having call failures or affecting the user experience The following standard metrics can help assess the QoS for signaling protocols Call setup time measures the time required to setup the call including the acknowledgement from the called party This metric has significance if the called party answers immediately In practice someone may answer after a variable time Post dial delay measures the time required to rece
83. IP SRC PORT DEST IP DEST PORT 192 168 1 100 5060 193 16 148 244 INVITE sip remoteserviceprovider com SIP 2 0 Via SIP 2 0 UDP 192 168 1 100 5060 From Alice lt sip 8184443118 myserviceprovider com gt tag 4000a5000 To Bob lt sip 9195642244 remoteserviceprovider com gt Contact Alice lt sip 8184443118 192 168 1 100 5060 gt Content Length 0 v 0 o Alice 2890844526 2890844526 IN IP4 192 168 1 100 s c IN IP4 192 168 1 100 t 0 0 m audio 10000 RTP AVP O 8 101 a rtpmap 0 PCMU 8000 Figure 125 Sample INVITE message with SDP offer VoIP NAT Traversal Issue Solutions The solutions can be classified as e Client based solutions e Server based solutions Client based solutions add intelligence to the endpoint that detects the IP address and port number used by the NAT translation applying translation within the application specific content Examples of such solutions include e Simple Traversal of UDP through NAT STUN e Traversal using Relay NAT TURN e Interactive Connectivity Establishment ICE Server based solutions assume a public server will resolve the translations specific to the application layer Solution examples include e Application layer gateway e Session border controllers PN 915 2611 01 Rev H June 2014 134 Test Case Using VoIP to Measure NAT PAT Performance STUN Simple Traversal of UDP Through NAT Defined by RFC 5389 STUN is a lightweight protocol that allows a wide varie
84. IP 2 0 UDP 192 168 1 100 5060 From Alice lt 31p 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt s1p 8184443118 myserviceprovider com gt Contact Alice lt s1ip 8184443118 192 168 1 100 5060 gt Content Length 0 Figure 122 IP Header and IP payload data for 2 REGISTER message after NAT traversal After this operation the NAT table includes the following information NAT TABLE after the REGISTER message passed the NAT ID SRCIP SRCPORT DESTIP DEST PORT 192 168 1100 75 83 202 16 Figure 123 NAT table after REGISTER sent through NAT We will assume that the REGISTRAR server is not behind a NAT ing device and responds with a 200 OK Upon receipt of REGISTER request the REGISTRAR server extracts the IP address and port number from the top Via header which includes the address of the last hop that sent the message and use it as a destination for the 200 OK response Because the Via header was not updated by the NAT it includes the private IP address 192 168 1 100 5060 which is not routable The structure of the response message is shown below SRCIP SRCPORT DESTIP DEST PORT 193 16 148 244 192 168 1 100 200 OK SIP 2 0 Via SIP 2 0 UDP 193 16 148 244 5060 From lt sip 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt 51p 8184443118 myserviceprovider com gt Contact Alice lt sip 8184443118 192 168 1 100 5060 gt Content Length Expires 180 Figure 124 Pack
85. IP addresses is used to map the test scenarios that execute the call flow for the registration requests Should a single IP address has to be used in the second range Channel mapping rules for SIP UA will be set to Use same value per port for IP address and UDP TCP TLS port will be set to Use consecutive values per port when a SIP Peer activity is used under a SIP cloud the emulated endpoints have to use distinct IP address UDP Port tuples 3 On the Terminate network associate each IP ranges with one of the activities To access the IP mapping configuration window click on Traffic and then on the tab IP Mappings PN 915 2611 01 Rev H June 2014 109 Test Case Determining the Maximum Transaction Rate for VoIP Protocols 4 Adda SIP peer activity on the Originate network and SIP Cloud and SIP Peer activity on the Terminate network Traffic Activities amp Endpoints Network Ranges By Port Distribution Group oIPSipCloud1 User_Agents El Groupi Consecutive IPs IP Mappings Command Editor o Network Range IP R3 in Private 20 1 101 1 8000 C rr1 IP Round Robin al fia Network Range IP F1 in Private 20 1 100 1 1 C C Figure 97 IP Mappings on Terminate network The IP ranges groups must be set to Round Robin for the SIP server VolPSipCloud and Consecutive IPs for the User Agents 5 Create the test scenario editing the following call flow on two channels Scenario Channel 0 Activity VolPSipPe
86. It is deployed on a large number of applications and technologies such as high definition content delivery websites terrestrial or satellite television Blu ray Discs or even online content stores The H 264 MPEG Part 10 or Advanced Video Coding AVC is developed to deliver video content at lower bitrate in regards to its predecessors without any increase in the design PN 915 2611 01 Rev H June 2014 219 Test Case Measuring Quality of Experience for Multimedia VoIP Calls complexity or cost of implementation An additional goal is to allow selection for bit rates scaling for high or low video resolutions and to allow broadcasting of content disk storage IP encapsulation and even telephony transport The main features promoted by the codec are e Multipicture inter picture prediction that allows significant bit rate reduction for repetitive motion content e Variable block size motion compensation that allows precise segmentation of moving regions increasing the efficiency over used bitrate e Multiple motion vectors per macroblock that allow a robust motion detection decoding at the player side e Motion compensation with weighted prediction e Lossless macroblock coding that allow representation of specific regions without consuming large amounts of data e Elaborated algorithms for spatial blocking transform optimization e Context adaptive binary arithmetic coding CABAC as highly data efficient encoder e Context adaptive variabl
87. Jitter was reported what is the Packets with Delay Variation Jitter up to 10 distribution of the Delay Variation Jitter S A values S S PN 915 2611 01 Rev H June 2014 186 Test Case Determining the Performance of a Session Border Controller Table 73 Distribution of RTP Consecutive Lost Packets Statistic Name Questions Consecutive Loss of One Packet Sequence Consecutive Loss of Two or Three Packet Sequences Assuming that packet loss was i reported what is the Consecutive Loss of Four or Five Packet Sequences distribution of the lost RTP Consecutive Loss of Six to Ten Packet Sequences packets Consecutive Loss of Eleven or More Packet Sequence Table 74 RTP Streams Statistic Name Questions EOneRHenh yk eaS Assuming that packet loss was reported what is the Concurrent RTP Streams distribution of the lost RTP Packets max Number of calls with incoming RTP packets Number of calls without incoming RTP packets Are any calls without RTP Are any calls with RTP Does this number match the number of Calls Connected 2 Conclusions This test methodology provided details of how to emulate with IxLoad Voice a SIP Registrar and Proxy Server to act as a Back to Back user agent This configuration is presented in the context of measuring the performances of a Session Border Controller The most challenging part of the configuration is setting correctly the dispatching rules for each activity PN 915 26
88. June 2014 19 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems b Inthe Activities column the SIP and RTP check boxes are selected Notes Did you notice the oF Create New Group Remove Selected Group eg Edit Modify Selected Distribution Group and Move Up Move Down buttons Try them and see how they work Traffici User Source IP Mapping Activity User Source IP Rule per port Make _Call Use Consecutive IPs Activities amp Endpoints Make Call Network Ranges By Port Distribution Group es amp IP 1 Groupi Consecutive IPs z a Network Range IP R 1 in Metwork1 20 1 1 1 8000 Advanced Settings IF Mappings Command Editor Figure 14 Traffic configuration example for SIP IP phone simulation Configuring Traffic 2 for Network 2 This section describes how to map distinct network ranges for SIP traffic and RTP traffic when simulating a network where SIP and RTP traffic have separate IP sources In our example the SIP server and the media gateway components are separate entities and they use separate IPs The steps highlight how the IP range 20 1 100 1 1 is associated with the SIP server component and 20 1 200 1 1 is associated with the media gateway component Traffic2 IF Mappings terre Network Ranges By Port Distribution Group Receive Call Endpoint Type E SIPProxy Consecutive IPs Network Range 1 in Networks 20 100 1 1 1
89. Mode Request Change bitrate to 0 6 60 kbps after C packets L Change bitrate to after packets _ Iterate for the entire call duration _ Respond on Mode Change requests Figure 190 Set the AMR WEB as preferred codec PN 915 2611 01 Rev H June 2014 201 Test Case Measuring Quality of Experience for Voice Calls in LTE e Set the Audio parameters To have media during an established SIP call the call flow must contain a media function and that type of media to be enabled on the activity The call flow contains the function Voice Session that plays and listen audio clips in the same type it implements the full duplex audio functionality On the Audio tab the Audio Activity has to be enabled In the same page we can specify the duration of the voice session and implicit the duration of the call assuming only voice session function is executed during the call Select the Audio tab enable the checkbox Enable audio on this activity set the Play for to 30 seconds and enable the Perform MOS checkbox Traffic VoIPSipPeer 1 VoIPSip Peer Scenario RIG Audio Enable audio on this activity if unchecked all audio script functions will be SKIPPED z Enable audi this activity if unchecked all aud t functi ill be SKIPPED Play Settings _ Enable jitter buffer Clip US_042 wav Format PCM Duration 32785 ms Size 524556 bytes Output level 20 dBm C Play for dip duration or TalkTime all objectives except Chan
90. Number User defined 1001 1010 a files Pool i Security Profles Pool ma _ File Proxy Security Settings Shared Secret Authenticate REGISTER Requests J 6 User defined Authenticate INVITE Requests File Authentication Algorithm Operator Variant User defined Enable AKA Authenticat O File AKA v1 AKA v2 Username Quality of Protection User defined user0 001 010 gt auth C authint File Registration Expiration Time 3600 seconds Realm ixiacom 0 com Password User defined 43345 l _ File Log Event Viewer OK Cancel Figure 195 Security profile on the SIP Server 15 Set the parameters of VolPSipPeer2 activity Select the VolPSipPeer2 activity PN 915 2611 01 Rev H June 2014 205 Test Case Measuring Quality of Experience for Voice Calls in LTE a Set the Dial Plan In this example we ll emulate 10 User Agents with Phone Numbers 2001 to 2010 select the Dial Plan tab and edit the sequence 2001 2010 in the Source Phone Numbers field of the Dial Plan Leave the Destination as none this activity is only receiving calls so it does not need destination phone numbers Traffic VoIPSipPeer2 VoIPSip Peer Source Destination IPs The source IP addresses are taken from the associated IPs None Network see Traffic Network mapping tables in the test Phone numbers L Override phone numbers from destination activity C Phone bo
91. Number of Concurrent Calls 4 Select the Enable Parallel H 245 check box Trafficl Make_Call YoIPH323 Peer Scenario Execution Dial Plan 23 J Terminal capabity Codecs RTP Aud vdeo Other Enable signaling on this activity iF unchecked all H323 script Functions will be SKIPPED H323 Specific Settings Advanced Signalling Options Bandwidth 4nd Terminal Type Enable FastStart Bandwidth fin Kbps 64 Enable Tunneling Terminal Type Terminal Entity without ME 50 Enable Parallel H245 Versions Send Call Alerting H225 versi 0 0 8 2250 0 5 Er _ Send Call Proceeding ARER H 245 version 0 0 8 245 0 9 Figure 65 Configuration example for H 245 in parallel with FAST START Results Analysis The following questions are provided to help discover performance issues more readily 1 Has the test objective been achieved Use the Throughput Outbound Inbound statistics and Concurrent RTP Streams statistics available in the RTP Streams view RTP Streams oIPH323 A E bP S R 8 lol 44 000 00 40 000 00 36 000 00 32 000 00 28 000 00 24 000 00 20 000 00 16 000 00 42 000 00 8 000 00 4 000 00 0 00 13 31 49 1 07 T2 1 43 2 01 219 23r Paaa 3 13 3 31 3 49 4 07 4 25 4 43 5 01 5 19 SPF 40 56 1 16 1 34 1 32 2 28 2 46 3 04 3 22 3 40 3 58 4 16 4 34 5 10 2 58 3 01 3 04 3 07 3 10 313 3 16 313 eae 3 25 3 28 IP VaPHE23 Number of Cals wih inconng RTP paces ooo oo cal om cao om ow oo oo oof
92. Offer Send custom message body Edit Custom Send a sip request message Restore Defaults Figure 230 Configuring expiration value over the permitted range PN 915 2611 01 Rev H June 2014 246 Test Case Telephony Denial of Service Params Behavior Flow Manager Extract Variables Authentication Output _4 gt REGISTER AUTO REQUEST URI SIP 2 0 Via AUTO VIA From AUTO FROM To AUTO ToO Call ID AUTO CALL ID CSeq AUTO CSEQ Contact AUTO CONTACT Max Forwards 70 Content Length AUTO CONTENT LENGTH Expires Sa9b9c9d9e9f Create From Template Load From File Edit Options Change case Modified by user W Case sensitive Delay before execution Static Expression fo mg Message body Send audio SOP Offer Send custom message body Edit Custom Send a sip request message Restore Defaults Figure 231 Configuring expiration header with alphanumeric characters PN 915 2611 01 Rev H June 2014 24 Test Case Telephony Denial of Service 12 Repeat steps from 4 to 10 to add a new procedure to the call flow and multiply the To From and Contact headers in the originating requests Additionally you may configure the header value to contain values from other variables Consult the example below IP Send Request Properti Params Flow Manager Extract Variables Authentication Output _4 r REGISTER AUTO REQUEST URI SIP 2 0 Via SIP Vial Via SIP From Via
93. P address of the emulated SIP Server 5 Click Traffic2 under SIP_Proxy NetTraffic and check whether each activity has the SIP traffic mapped to one and only one network range Check that the Distribution Group for the range used by SIP_Proxy activity the cloud is Round Robin EE EE t i le seeene Caller ee acm Callee m Trafficz User Source IP Mapping Activity User Source IP Rule per pork R Caller Use Consecutive IPs Command Editor Activities amp Endpoints Network Ranges By Port Distribution Group SIP_Proxy R_ Caller R_Callee E IP 3 Groupi Consecutive IPs IP Mappings 7 m LJ L m m L m E Network Range IP R4 in SIP_Proxy 40 40 100 1 1000 ia isa l Ll Ll Network Range IP R5 in SIP_Proxy 40 40 110 1 1000 ia ia L Ll L F Network Range IP R6 in SIP_Proxy 40 40 120 1 1000 P i liii L Network Range IP R in SIPF_Proxy 40 40 130 1 1000 P a i L L l m m m iial L m Network Range IP R3 in SIP_Proxy 40 40 50 1 1 Ll m an m Ll L Figure 166 SIP activities mapping to IP ranges for emulated SIP Proxy PN 915 2611 01 Rev H June 2014 176 Test Case Determining the Performance of a Session Border Controller 6 Click the SIP_Caller NetTraffic and change the network settings accordingly to the particular test topology The number of IP addresses should match the number of channels defined in Test Objective 1000 in this example 7 C
94. Proxy activity the Cloud settings PN 915 2611 01 Rev H June 2014 179 Test Case Determining the Performance of a Session Border Controller Setting the R_Caller Activity Parameters To emulate the Registrar server the IxLoad conjuration has to contain a SIP Peer activity waiting for SIP Register messages and send the proper response In addition this activity extracts the SIP_Contact information from the incoming Register message and passes it to the activity that handle the call setup There are two activities for Registrar emulation one for each set of SIP User Agent one for SIP Callers and one for SIP Callees Each activity has its dial plan match the dial plan of the SIP User Agents 1 Click the R_Caller activity under SIP_Proxy NetTraffic 2 Check the Scenario it contains a procedure waiting for SIP Register and sending the response 200 Ok It can be modified to emulate the Register with authentication enabled by adding script objects functions to send 401 Unauthorized and wait for the second Register 3 Click the Dial Plan tab and set the Source Phone Numbers 5557 0000 in this example this sequence should match the one specified as Source Phone Numbers for the SIP_Caller activity this is the criteria for matching the incoming SIP Register messages to this activity 4 Click the Cloud tab and ensure that the Enable SIP Cloud simulation using setting from check box is selected 5 Check whether the dispatching rule in
95. Questions Have the channels been connected continuously Active channels at a constant rate during the Sustain Time Successful channels Are there any failed channels identified Failed channels A What is the quality of the audio streams transmitted Check the RTP MOS VolPSip view Table 80 Statistic Name Quality Questions MOS Best Are all values above 4 MOS Worst i a Were there any streams with less than 4 score MOS percallbest o oo MOS per call worst lo ooo o OSO What is the quality of the video streams transmitted Check the Video RTP Relative MOS VolPSip view Table 81 Statistic Name Value Quality Questions Average Minimum than 4 score Fa a S Maximum a O Average PN 915 2611 01 Rev H June 2014 231 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Check the Video RTP Absolute MOS VoIPSip view Table 82 Statistic Name Value Quality Questions Average o Were there any streams with less Minimum than 4 score Instant Absolute MOS V Maximum Completed Absolute MOS_V Average Reference MOS to user perceived call quality Table 83 MOS Value Quality Perceived Impairment Excellent Imperceptible Imperfections can be perceived Slightly annoying Nearly impossible to communicate Impossible to communicate Test Variables Use each of the following variables in separate test cases Use the above test case as a baseline and modify a few para
96. Registration Admission and Status RAS _ Enable RAS Retry and Timeouts For RAS Messages Maximum Petry Count Timeout fin secs Gatekeeper 4d a Tat Es Frc r SK Disco 1 931 User User 618501 1000 _ Hexadecimal Byte Stream Display xia 818501 1000 Figure 51 Configuration example for H 323 settings of Make_Call activity H 323 Normal Start Configuring the CODEC Settings Page for the VoIPH 323Peer1 Activity The following figure highlights the default settings for this page The Codec page controls the CODECs supported by the simulated phone the preferential order of the CODEC and additional CODEC options such as packet time Traffic Make_Call YoIPH323 Peer Audio Codecs Settings Payload type TT i feel Law 6711 Law Gubgoing payload type 0 Facket time 20 ms 160 bytes per frame Incoming payload type 0 ITU T G 711 is a standard to represent 6 bit a WARNING The outgoing payload type number compressed pulse code modulation PCM samples For i will be used for RTP header payload type when signals of voice Frequencies sampled at the rate of sending packets and viceversa 8000 samples second Please note that the outgoing payload type of a G 711 encoder creates a 64 Kbps bitstream sender must be equal to the incoming payload type of a receiver Figure 52 Configuration example for the Codec Settings of the Make_Call The configuration shown uses the default settings that negotiate G 711 uLa
97. SRC PORT 80 Figure 115 Incoming connection HTTP response To better explain the difference between NAPT mode and basic NAT mode we need to add a secondary host in the private network lets say 192 168 1 101 and see what happens when a single public IP address 75 83 202 16 is configured on the public interface of the NAPT enabled router Let us also assume the second host initiates a telnet connection from port 2500 to an external telnet server 207 46 197 32 which listens on port 23 for telnet connections If the first PN 915 2611 01 Rev H June 2014 127 Test Case Using VoIP to Measure NAT PAT Performance connection is still active basic NAT will simply not work because a secondary public IP address must be With NAPT the router can reuse the same public IP address 75 83 202 16 but a different port other than 32 000 which was used in the first connection Telnet Client 192 168 1 101 2500 initiates an outgoing connection to an external Telnet Server 207 46 197 32 Private Network Internal l Public Network External Il NAT NAPT 192 168 1 101 24 SRC IP 192 168 1 101 SRC PORT 2 500 DST IP 207 46 197 32 SRC IP 75 83 202 16 DST PORT 23 192 168 1 1 24 1 SRC PORT 2 500 DST IP 207 46 197 32 DST PORT 23 75 83 202 16 22 Figure 116 Outgoing Telnet client connection After the router receives the response packet it will perform a lookup in the NAT table and strip off the destination IP addr
98. Serrage 5 _ Override message timeout specified in scenario Timers REGISTER Expires Headers 3600 FETE Ean ree an with 64 71 For all responses and ACK Ti 500 wf ession Expirels z S UPDATE Session Expire s 90 a ean T2 4000 Ignore received retransmissions Retransmit ACK Figure 35 SIP Retransmissions settings Configuring the Codecs Settings for the Receive_Call activity 1 Click the Codecs tab 2 Make G 711 uLaw the preferred CODEC by moving it in the top of the Audio Codecs list its position in the CODEC list to 7 3 Set the RTP Packet time to 20 ms 160 bytes per frame Traffic Receive Call VoIPSip Peer Audio Codecs Packet time 20 ms 160 bytes per frame Incoming payload type 0 EEN al a i Outgoing payload type ITU T G11 is a standard to represent amp bit WARNING The outgoing payload type number compressed pulse code modulation PCM samples For D will be used For RTP header payload type when signals of voice Frequencies sampled at the rate of sending packets and viceversa S000 samples second Please note that the outgoing payload type of a 5 711 encoder creates a 64 Kbps bitstream sender must be equal to the incoming payload type of a receiver Figure 36 Configuring CODECs settings for the Receive_Call activity PN 915 2611 01 Rev H June 2014 34 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring Audio Settings for the R
99. Session Border Controller Table 64 Call Rate statistics Statistic Name Questions Have the calls been attempted continuously at a constant call rate during the Sustain Time Calls Attempted per Second Calls Connected per Second How do the Calls Attempted rate and the Calls Connected rate compare to each other 2 Have any call failures been reported Check the Calls view Table 65 Call statistics Statistic Name Questions 1 Have any call attempts failed Compare a Calls Attempted and Calls Received with b Calls Attempted and Calls Connected 3 Have any scenario loop failures been reported Check the Loops statistics view Table 66 Statistics highlighting the pass fail result based on call flow execution Statistic Name Questions Total Loops 1 Are the Successful Loops and Total Loops values equal 2 Have any Failed Loops Aborted Loops or Warning Loops Successful Loops been reported Failed Loops Aborted Loops Note failed aborted and warning loops highlights failures at the Warning Loops scenario level 4 Have all messages received by cloud been dispatched correctly Table 67 Dispatched messages stats Statistic Name Questions 1 Is any SIP message received and not dispatched Note If there are Undispatched messages the dispatching rules do not cover all the cases The Dispatching Rules set on each activity should be checked typically the error is in mismatching of the phone numb
100. Settings Advanced Signalling Options Bandwidth 4nd Terminal Type Enable FastStart Bandwidth fin Kbps Bt E Terminal Type Terminal Entity without Mc 50 _ Enable Parallel H245 Versions Send Call Alerting H 225 Versi 0 0 8 2 50 0 5 Send Call Proceeding ersion H 245 Version 0 0 6 245 0 9 kK E Figure 64 H 323 Configuration example for FAST START Configuration Highlights for H 245 in Parallel with Fast Start Another method used to speed up the negotiation of audio video parameters is defined in H 323 version 4 The method uses binary embedding in which the calling endpoint creates its H 245 request terminalCapabilitySet at the very beginning of the call and embeds this message in the H 225 0 Q 931 message setup In this way the called party knows the caller s entire capability list right from the beginning H 323v4 allows H 245 to start in parallel with fast connect by including H 245 messages in the Setup message By starting H 245 early two endpoints can establish an H 245 session faster in the event that the called endpoint rejects fast connect To configure the H 323 activities to use H 245 in parallel with fast start repeat the following steps for both activities 1 Click the H 323 configuration page of Make_Call 2 Verify that the Enable signaling on this activity check box is selected 3 Select the Enable FastStart check box PN 915 2611 01 Rev H June 2014 68 Test Case Determining the Maximum
101. Si u 1 DistGroupi VoIPSkinny yand Editor Figure 184 Add the VoIPSIP Peer activity 11 Add a SIP peer activity on the Terminate side Repeat the operation of adding a SIP peer activity on the terminate Net Traffic Terminate Figure 185 SIP Peer activities on both Originate and Terminate Network Traffics PN 915 2611 01 Rev H June 2014 197 Test Case Measuring Quality of Experience for Voice Calls in LTE 12 Use one of SIP Sample message flows as the call flows for VolPSIPPeer1 and 2 activities Drag and drop the lollipop of the VolPSIPPeer1 over VoIPSIPPeer2 Select the SIP EP Registration and Call with Voice message flow from the drop down list e ee Originate Figure 186 Traffic2 n F iv VolPSipPeer2 v Please select one of the sample message flow lt Default Test Scenario gt SIP EP Basic Call with Voice SIP EP Basic Call with Multimedia SIP EP Basie Call with T38 A EP Registration and Call with Voice SIP EP amp SIP Server Registration SIP EP amp SIP Cloud Registration and Make Call with Voice SIP EP amp SIP Cloud Registration and Receive Call with Voice Add SIP Call flow to the SIP activities 13 Set the parameters of VolPSIPPeer1 activity Select the VolPSIPPeer1 activity a Set the Dial Plan In this example we ll emulate 10 UEs with Phone Numbers 1001 to 1010 Select the Dial Plan tab and edit the sequence 1001 1010 in the Sourc
102. Started Templates VoIPSip Networks and Traffic Mew Traffic Flow apy E Originate aie DUT d Terminate ef ope Traffict cee E Traffic2 le Edit Replace With Loops Check Commands K Pi sr Makecall 41 O o o Error Ejo E Woice Session 1 OE Error eS Ri sip End Call Initiate d OK Figure 10 Overview of the IxLoad configuration sample Configuring the Network Parameters for Network1 Note This network hosts 8 000 SIP IP phones SIP clients Each simulated phone will use a distinct IPv4 IP address and a unique MAC address Select Network1 to display the IP network ranges 1 Set the following parameters a Stack IP over MAC VLAN PN 915 2611 01 Rev H June 2014 17 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems b IP Type IPv4 c Count 8 000 d Address 20 17 1 1 e Mask 8 f Gateway 0 0 0 0 Table 1 Summary of Network1 parameters IP Count Address Mask Increment Gateway Gateway MSS RX Type Increment 8 000 20 1 1 1 8 0 0 0 1 0 0 0 0 0 0 0 0 Stack 1 Filter 1 aoa pa Settings 1 Ep GratARP 1 GratARF 1 Dsi DNS 1 Ethernet 1 Unconfigured Figure 11 Network1 network configuration example for SIP IP Phones simulation Notes If the number of MAC addresses is higher than the maximum supported by the DUT add an emulated router right click the IP stack Insert below Emulated Router
103. T 3478 SRC IP 209 132 176 30 SRC PORT 3478 DST IP 192 168 1 102 DST PORT 5060 192 168 1 1 24 DST IP 75 83 202 16 DST PORT 1025 Figure 127 STUN Binding Response traveling through NAT e The NAT gateway receives the STUN binding response translating the destination IP address and port number from the IP header back to the address and port used by the STUN client when it initiated the STUN binding Request Further the application in our example the SIP client with STUN support uses the public IP address and port from the MAPPED ADDRESS field of the STUN binding response to construct the payload of the SIP requests initiated by the SIP client The mapping between the private IP port address and the public IP port address is typically deleted after a timeout when no traffic passes through the NAT To avoid removal of the NAT entries the STUN client periodically sends the STUN binding request to refresh the mapping on the NAT gateway Symmetric NAT prevents STUN from working due to the rules it uses to create the pinhole For non symmetric NAT the same pinhole will be used whenever the same endpoint from the private network sends a packet to a different destination With symmetric NAT a new pinhole will always be created each time the destination of the packet is changed regardless if the source remains the same With STUN the client sends the binding request packet and the STUN server replies with the public
104. T types e Full Cone NAT All requests from the same internal IP address and port are mapped to the same external IP address and port By sending a packet to the mapped external address any external host can send a packet to the internal host e Restricted Cone Uses the same IP and port mapping as a full cone NAT but unlike it an external host with IP address IP1 can send a packet to the internal host only if the internal host had previously sent a packet to IP address IP1 e Port Restricted Cone Port restricted cone NAT is similar with a restricted cone NAT but the restriction includes port numbers rather than an IP address For example an external host can send a packet with source IP address IP1 and source port P1 to the internal host only if the internal host had previously sent a packet to IP address IP1 and port P1 PN 915 2611 01 Rev H June 2014 129 Test Case Using VoIP to Measure NAT PAT Performance Symmetric NAT All requests from the same internal IP address and port to a specific destination IP address and port are mapped to the same external IP address and port If the same host sends a packet with the same source address and port but to a different destination a different mapping is used Furthermore only the external host that receives a packet can send a UDP packet back to the internal host NAT Capacities and Performance The capacity of the NAT table depends on the amount of memory that it has availab
105. Variables Test Tool Variables Table 41 Parameter Name Current Value Additional Options PN 915 2611 01 Rev H June 2014 116 Test Case Determining the Maximum Transaction Rate for VoIP Protocols DUT Test Variables Table 42 Parameter Name Current Value Additional Options IPv6 Transport Protocol SIP UDP SIP TCP SIP TLS Authorization Enabled Off Extension to Other Transaction Types Using the Loops per Second test objective it is possible to measure the DUT s capacity and rate for any type of SIP transaction For example to test the instant message rate only a few changes are required e The call flow should implement that shown in Figure 93 Instant messaging transaction e The dial plan must be changed on the emulated endpoints and emulated server In the emulated endpoints dial plan Originate network traffic the destination IP addresses can use the symbolic link to the VolPPeerSIP on the Terminate network traffic This dial plan can be different from the dial plan of the emulated endpoints while the test simulates sending messages from one endpoint over a proxy to other endpoint All other settings may remain the same In this case one execution loop is one transaction for instant messaging so the test objective Loops per Second is equivalent with Instant Message Transactions per Second Conclusions To assure a high QoE in a VoIP network it is important to assure that the network not only posses
106. _ Registrar server Figure 25 SIP Settings for back to back configuration Top Sample configuration with a SIP Server 61 15 1 1 acting as an Outbound Proxy Bottom Configuring Automatic Behavior for the Make_Call Activity 1 Click the Automatic tab 2 Select the Enable Retransmission Ignore received retransmissions and Retransmit ACK check boxes and leave the default values for the T1 and T2 timers a T1 500 ms default b T2 4000 ms default PN 915 2611 01 Rev H June 2014 28 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Scenario Execution ial Pan 517 Automatic Tis coud codecs RTP Audio _ Enable session timers Fax 30 ERT Other usj a Yale A AE TETEE 1 E CxDIraCGO Timeout and retransmisions Clon ye per wpe gf message eS Override message timeout specified in scenario Timers REGISTER Expires Headerts 3600 INVITE Seka at with 64 T1 For all responses and ACK Ti 500 e ession Expirels UPDATE Session Expiret s 90 Enable retransmissions T2 4000 Ignore received retransmissions Reetransmit ACK Figure 26 SIP Retransmissions settings Configuring CODECs Settings for the Make_Call Activity 1 Click the Codecs tab 2 Make G 711 uLaw the preferred CODEC by setting its position in the CODEC list to 7 3 Set RTP Packet time to 20 ms corresponding with a payload of 160 bytes per frame Audio Codecs E p Facke
107. a distributed application layer gateway used as a media border element performing trans coding G 711 codec is used in the private network and G 729 in the public network PN 915 2611 01 Rev H June 2014 89 Test Case Subjective Quality of Voice Step by Step Instructions The final IxLoad configuration as a result of these steps is provided on blackbook ixiacom com Web site see xLoad 5 10 Voice PESQ crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test 1 Configure a test that uses IxLoad Voice endpoint pairs to simulate SIP calls with media Scenario Channel 0 Activity VoIPSipPeer Traffic Traffict Hetwork1 Column Originate Link VoIPLink1 USIP Recei Error Figure 75 Test scenario to establish SIP calls with media Use the instructions for Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems listed in this booklet or refer to the IxLoad user manual for network configuration test scenario editing timeline establishment objective setting and port assignment The test is configured with two networks each with 1 500 distinct static IP addresses The SIP VolPPeer protocol is used on both networks and the test objective is 1 500 channels
108. a parameter This is particularly useful when you wish to force a number of loops per channel forcing deterministic results The Sustain Time must be set to number of loops x Talk Time 10 seconds or more 4 Select the Graceful Ramp down check box this is the default setting PN 915 2611 01 Rev H June 2014 30 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Traffic Receive _Call VoIPSip Peer Run For Loop delays Aliases 2 the entire test duration Before 1st loop Oo ms Number of aliases phone numbers per channel 1 C a number of loops Between loops 0 ms NOTE IF more than one aliases will cycle Channel mapping rules For SIP UA Channel mapping rules For media IP address Use same value per port IP address Use same value per port UDPITCFITLS port Use same value Port Use consecutive values per port Phone ma Use consecutive values per port VeriFy all settings Restore defaults Figure 30 Configuration Example for the Execution Setting page of the SIP Receive_Call activity 5 Set the Channel mapping rules for SIP when sharing the same SIP IP address and SIP port number for all the channels a IP Address Use same value per port Note This mode requires a unique RTP port for each simulated channel The RTP port is specified as a range using a sequence generator 10000 65534 4 and can be configured using the RTP settings page see Step 7 below
109. a rtpmap 9 G 22 8000 a rtpmap 0 PCMU 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv Figure 206 Sample Audio capabilities advertised by single screen endpoint For the video codecs the options are complex and the SDP information can become quite elaborate In the example below the endpoint advertise the H 264 as initial primary video codec with alternate options of H 263 codec The below image has been intentionally truncated to fit the page layout m video 20002 RTP AVP 91 93 94 b TIAS 5000000 a rtpmap 391 H264 90000 a fmtp 91 profile level id 42801 6 max br 5000 max mbps 30000 max fs 3600 max smbps 1 08000 max fps 6000 max remd nalu size 1382400 a rtpmap 93 H263 1998 90000 a fmtp 93 custom 1280 768 3 custom 1 280 20 3 custom 1024 768 1 custom 1024 576 2 custom 800 600 1 cif4 1 custom 20 480 1 custom 640 480 1 custom 51 2 288 1 cif 1 custom 352 240 1 qcif 1 maxbr 7680 a rtpmap 94 H263 90000 a ftmtp 94 cif4 1 cif 1 gcif 1 maxbr 7680 a rtcp fb nack pli a rtcp fb ccm fir a rtcp fb ccm tmmbr a sendrecv a content main a answer tull Figure 207 Sample Video capabilities advertised by single screen endpoint The main protocol for video in telepresence is H 264 MPEG 4 This represents the standard for video compression of captured media that requires good quality transmission of high definition video content The codec is developed by International Telecommunication Union ITU
110. action Rate for VoIP Protocols Test Case Determining the Maximum Transaction Rate for VoIP Protocols Overview With respect to VoIP bandwidth consumption signaling protocols only account for a small fraction of the total traffic with the media segment consuming most of the bandwidth This often results in network design that maximizes subscribers QoE based solely on media requirements Signaling requirements however play a crucial role in the call setup process impacting subscribers QoE as much as media Signaling protocols are used mainly to establish calls between two or more endpoints The associated performance metric is the CSR and is measure in calls per second CPS This is covered in Test Case Determining the Maximum Call Setup Rate CPS in this booklet Signaling protocols are used for other purposes as well they need to be designed into networks to ensure positive QoE and avoid service interruption With respect to the messages exchanged during a SIP call six signaling messages are used with a total payload of approximately 40kb For a call duration of 3 minutes using a G 711 CODEC 64Kbps the signaling represents less than 1 percent of the VoIP traffic Alice s Bob s softphone SIP phone INVITE 180 Ringing 200 OK ACK lt Both Way RTP Media gt BYE 200 OK Figure 89 Basic SIP call flow Most VoIP networks require that devices register before being able to originate or terminate a call The registration
111. address assigned to their router Network Address Port Translation NAPT The NAPT mode is also Known as port address translation PAT IP masquerading or as NAT overload The different names come from the way NAPT NAPT is the most common method used The difference between NAT and NAPT is quite easy to remember NAT translates IP addresses the source IP for outgoing connections and destination IP for their responses while NAPT translates the ports NAPT includes a mix of static NAT and dynamic NAT but with a significant enhancement port address translation support NAPT takes a static or dynamic IP address that is bound to the public interface of the NAT enabled router and allows all hosts within the private network to access the public network PN 915 2611 01 Rev H June 2014 126 Test Case Using VoIP to Measure NAT PAT Performance NAPT Explained Assume a host on a private network has an IP address of 192 168 1 100 which sends an HTTP GET packet to a public HTTP server with the IP Address 209 132 176 30 As the original packet passes through the NAPT enabled router the source IP address field is changed by the router from 192 168 1 100 to 209 132 176 30 However because this was the first and the only connection active the source port number field is not changed This is better illustrated in the following figure HTTP Client 192 168 1 100 32 000 initiates an outgoing connection to an external HTTP Server 209 132 176 30
112. affic is full duplex and the quality of voice is evaluated at both ends the number of channels with PEs Traffici Makecall VoIPSip Peer Enable audio on this activity iF unchecked all audio script Functions will be SKIPPED Play Settings Enable jitter buffer Clip US_042 way Buffer size 100 ms Format M4 Use compensation Output level 20 dBm oe E ns G Play for clip duration or TalkTime all objectives except Channels Max dropped consecutive packets C Play for Perform Qoy Type OF Service Units of Channels TOSIDSCE Class 1 0x20 value 150 Channel Selection Evenly Spaced Channels Perform MOS Calculate One Way Delay _ Generate silence Figure 78 Audio settings 4 Analysis is set to 750 In this example the test objective is set to 7500 channels with the Evenly Spaced Channels option selected which will result in analysis of every tenth channel 1500 divided by 150 Similar settings should be applied to both activities call origination and termination The number of channels that can be analyzed depends on the resources available PN 915 2611 01 Rev H June 2014 93 Test Case Subjective Quality of Voice If insufficient resources are available to analyze all the calls the subset should be selected based on the expected behavior of the DUT e Evenly or random distributed if no errors are expected on the DUT or if the errors occur randomly e In blocks if errors are expected on a speci
113. ality of Service in Converged Networks Results Analysis Analysis is focused on VoIP measurements Beside the quality of voice metrics the most important statistics are video and data traffic throughput Check the Video Client Data Rates view and compare the obtained values with the expected ones Check the HTTP Client Throughput Objective values for this traffic 1 Client Data Rates sa olx S p l a BA 16 000 14 000 12 000 9 20 9 40 10 00 10 20 10 40 9 30 950 10 10 10 30 10 50 Figure 71 Video Client Data Rates and HTTP Client Throughput Objective PN 915 2611 01 Rev H June 2014 84 Test Case VoIP Quality of Service in Converged Networks To verify the QoS for VoIP the call completion statistics for signaling and MOS values for media are the most important Channels VolPSip Call Rates WoIPSip EME 5 14 5 46 4 18 4 50 10 22 5 38 7 10 7 42 Bile B 46 9 18 3 9 50 etre a t a eye gia ile e tele Cece leigh linda sol sis J PETAn fo td EEJ ago woz 4 Stat Name MoS Instant Avgi 1 600 MOS Instant Best MOS Instant Worst 4 410 4 410 4 410 4 410 1 200 Mis Besk 4 410 4 410 4 410 4 410 MOS Worst 4 410 4 410 4 410 4 410 M S Per Call avg 4 410 4 410 4 410 4 410 POS Per Call Best 4 410 4 410 4 410 4 410 10 50 MOS Per Call Worst 4 410 4 410 4 410 4 410 E gt Figure 73 Calls VoIPSIP and RTP MOS VoIPSIP In this example there a
114. art function using the symbol Trafficl VoIPSipPeer 1 VoIPSip Peer Scenario Execution D HA b H x oO Tm Evcheek Se a a ef HE B Add Channel H Note Zoom 100 Workspace A Scenario Editor lt TestScenario1 gt _ SIP IMS EndCall Receive ass Scenario Channel 0 SIP IMS MakeCall SIP IMS MakeRegistration SIP IMS ReceiveCall SIP IMS Subscribe 4 SIP MakeCall Authentication SIP MakeCall Complete 4 SIP MakeCall Redirect Server SIP MakeCall Route 4 SIP MakeCall SIP MakeRegistration Authentication m SIP MakeRegistration Complete SIP MakeRegistration First Loop Only Figure 226 Adding a predefined procedure in the Scenario Editor PN 915 2611 01 Rev H June 2014 241 Test Case Telephony Denial of Service 5 Double click on the SIP MakeRegistration Authentication procedure in the Scenario Editor to access the containing procedures The following steps will detail the actions needed to modify the SIP signaling headers to alter the proper behavior of an SIP endpoint and exploit any issue that might cause a security fault of the DUT 6 This new view will allow access to all the predefined call flow messages Open the first Send REGISTER procedure and modify Contact header information to point to 127 0 0 1 as the localhost address The expected behavior of the DUT is to register the originating phone number in its database as a local res
115. as a result of these steps is provided on the blackbook ixiacom com Web site see IxLoad 5 10 Voice SIP to PSTN crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test Create the Network for SIP Activity 1 Open the IxLoad GUI 2 Adda NettTraffic activity and select the network Network 1 3 Set the network parameters as shown in the following table Table 55 Summary of Network 1 parameters IP Count Address Mask Increment Gateway Gateway MSS RX Type Increment 192 168 1 101 0 0 0 0 0 0 0 0 Add and Configure the SIP Peer Activity 1 Add a VoIPSIP Peer activity 2 Rename the new added activity from VolPSipPeer1 to SIPMakeCall 3 Edit the Scenario by adding the SIP Make Call Complete procedure the Voice Session function and the End Call Initiate procedure TIP an existing test scenario may be loaded instead of creating it from scratch For example you may load VS_022 DUT_S Pv4 MakeCall EndCall with RTP 33s tst provided with the product PN 915 2611 01 Rev H June 2014 157 Test Case Determining the Capacity of a VoIP to PSTN Gateway 4 Click the Dial Plan tab and configure the Destination phone numbers to user defined values 717444 0001 0030 to modify the user defined value
116. ate and change the call flow in the Scenario Editor or the IxLoad VolP User Guide for instructions of how to modify the call flow 3 Click the Dial Plan tab and set the Source Phone Numbers 5552 0000 in this example this sequence should match the one specified as Destination Phone Numbers for the SIP_Caller activity 4 Click the SIP tab and set the Server Address to the IP address of the SIP Proxy server or the IP address of the DUT private side depending on the type of DUT 5 Click the Audio tab and ensure that the Enable audio on this activity check box is selected this activity will play and receive audio on the established calls Setting the SIP_Proxy Activity Parameters SIP_Proxy is an activity of type SIP Cloud It listens on the specified port 5060 in this example and dispatches the incoming SIP messages to one of the SIP Peer activities mapped to it 1 Click the SIP_Proxy activity under SIP_Proxy NetTraffic 2 Inthe Settings tab set the IP Address to the network range IP R3 the one with Distribution Rule Round Robin see the section Setting the SIP_Caller activity parameters The UDP port is 5060 change it if needed d 2 SIP_Caller i SIP_Prony ae ae gb i Traffict Eel fal caller SIP _ Callee gP Traffic3 yaffice SIP_Proxy YoIPSipCloud Settings Preview Cloud Traffic IP Preference Only IP C Only IP 6 060 ixload com Figure 169 SIP_
117. atic tool thus preventing any other incoming or outgoing calls These can look perfectly legitimate from signaling point of view and might represent a prank call or a form of telephone harassment However the originator can have a spoofed caller identity which makes the tracing even more complicated Spoofed identity just like in the source IP DoS amplifies the level of security exposure and typically ends in having blacklisted an entire range of legitimate phone numbers These types of organized telephony DoS TDoS might be triggered by social networks groups that want their point of view to be heard by the greater public or they react to an action initiated by the target of the attack These are typically towards financial institutions governmental agencies public sector companies or even political campaign offices or media broadcasting corporations There has been a world wide increase of such target specific telephone attacks and agencies such as Department of Homeland Security and the FBI are taking actions to prevent and eventually pursue complaints received from individuals or companies The most noticed forms of TDoS were fraud call for debt collection and extortion scams The attacker demands a sum of money and failing to receive this amount will trigger a flood of streams clogging the system for a long period of time A newer threat observed is the malicious signaling traffic intended to corrupt or compromise telephony systems deplo
118. ation helps to guarantee required bandwidth while minimizing latency jitter and packet loss Minimizing those effects is assured by detecting and prioritizing voice traffic thus controlling available bandwidth Data traffic may be impacted when voice traffic is prioritized because it is restricted from accessing bandwidth when VoIP needs it However data applications are less sensitive to packet loss and delay and user experience is less affected PN 915 2611 01 Rev H June 2014 4 Voice over IP Proper testing of device handling of both data and voice traffic requires generating a mix of data and voice traffic that is common in the environment s where the DUT will be deployed Baseline performance tests must be conducted for each individual protocol passing through the network Complexity In a distributed architecture in which devices from multiple vendors are used VoIP networks add an additional layer of complexity Interoperability between different protocols for example SIP to H 323 H248 to SIP and the interaction between different media for example PSTN wireless and pure VoIP networks can lead to additional issues related to protocol translation for example SIP to H 323 and multiple media encoding decoding operations that lead to degradation of speech quality Complex Features When VoIP shares a common network with data applications unified services and extended feature sets can be provided beyond those available with
119. attempted call rate is not constant has large variations Call rate is constant but the calls are attempted at a lower rate than configured in the test objective PN 915 2611 01 Rev H Table 11 Troubleshooting Solution Check the SIP Retransmissions counter Call Setup Time and End Call Time measurements A high number of retransmissions is an indication that the SUT cannot maintain the load generated This leads to larger call setup and teardown times which affects the number of users available to place new calls Increasing the Estimated Overhead Time parameter of the Calls Initiated per Second objective can force the tool to maintain a higher number of users available to place calls by decreasing the value of the estimated Call Hold Time Talk Time Hence the call duration will be shorter and the user will become available faster This issue is common when the Talk Time set in the script is hardcoded to a value larger than the estimated Talk Time provided by the test objective Correct the problem by configuring the media script objects to play for the Talk Time duration calculated by the test objective or use a hardcoded value lower than the one estimated by the test objective Verify that the Call Rate Attempted does not exceed the published performance Ixia provides several cards that may have a lower or higher performance Increase the Estimated Overhead Time value exposed by the Calls Initiated per Second objecti
120. authentication methods are defined for SIP IxLoad supporting all of them In this example we ll use digest MD5 that requests each user to have a username and a password We will use the sequence user0001 to user0010 for the username and the same password 12345 for all users it is possible also to use a sequence for the password also In cases when the phone numbers usernames or passwords of the emulated UEs are not in sequence a phonebook can be used In the SIP tab edit the Username and Password fields Trafficl VoIPSipPeer 1 VoIPSip Peer ial Gn SIP ABamatic Enable signaling on this activity SIP Port 5060 if unchecked all SIP script functions will be SKIPPED Transport settings Use external server VoLTE settings Maximum message size on UDP 1024 Server address 99 93 9731 Enable Call Control of dedicated bearer L Override transport specified in scenario Server port 5060 Domain name or local IP 22 22 22 1 _ TCP send immediate E Ouni proncy Enable FQDN resolution Registrar server _ Auto register simulated user agents Authentication UAC P Override registrar User names user0 001 010 Construction of SIP messages Password dzs gt AKA authentication settings ia Override default contact settings Select configuration lt None gt _ Override default destination domain name or host port Edit configurations _ Use Tel URI scheme for Source L Type Of Service _ Use Tel URI scheme
121. ce and the requirements for bandwidth it is useful to capture the SDP information or extract it from endpoints configuration if available The expected structure of SDP headers for teleconference call is the following e P address that is going to be used for the media transactions e The audio codecs supported o the preference of use o the details about coding time and o required bandwidth for each of them e Video codecs supported o the preference of use o the details about encoding the media as frame rate encoding mode video resolution o maximum bitrate and the RTCP details that assists real time media negotiation e Optional video codec for presentation channel that can encode the delivered content to the User Interface e Optional application interactivity for content delivery These optional channels enhance the audio visual The use of such application depends on the technical capabilities of the endpoints PN 915 2611 01 Rev H June 2014 218 Test Case Measuring Quality of Experience for Multimedia VoIP Calls m audio 10000 RTP AVP 100 102 103 104 10590101 b TIAS 64000 a rtpmap 100 MP4A LATM S90000 a fmtp 100 profile level id 24 object 23 bitrate 64000 a rtpmap 102 MP4A LATM S0000 a fmtp 102 profile level id 24 object 23 bitrate 56000 a rtpmap 103 MP4A LATM 90000 a fmtp 103 profile level id 24 object 23 bitrate 48000 a rtpmap 104 G7221 16000 a fmtp 104 bitrate 32000 a rtpmap 105 G7221 16000 a fmtp 105 bitrate 24000
122. configuration as a result of these steps is provided on blackbook ixiacom com Web site see IxLoad 5 10 Voice Registration per Second crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test PN 915 2611 01 Rev H June 2014 108 Test Case Determining the Maximum Transaction Rate for VoIP Protocols Open the Configuration Template Start the IxLoad GUI 2 Create and configure two NetTraffics In this example the networks are named Public and Private Set 8000 IP addresses that will be used for emulated endpoints on the Originate side and 8001 IP addresses on the Terminate side i aha Prive aby S Traffic3 de S Traffic2 Ew vorrsipPeert a Public F l2 at IP 3 mac vian 3 f Fi Foma Unconfigured 8000 0 0 0 0 Increment every subnet Figure 95 Originating network IP range om IP R1 Unconfiqured IPv4 20 1 100 1 16 0 0 0 1 1 0 0 0 0 0 0 0 0 Increment every subnet IP R3 Unconfigured IPv4 20 1 101 1 16 0 0 0 1 8000 0 0 0 0 0 0 0 0 Increment every subnet Figure 96 Terminating network IP ranges On the Terminate network the first range with 1 IP address defines the IP address of the emulated SIP Registrar sever The second range with 8000
123. cted 5 Check whether the dispatching rule in Override default dispatching rules has the proper sequence of phone numbers in the Formula for dispatching field 5552 0000 in this example PN 915 2611 01 Rev H June 2014 181 Test Case Determining the Performance of a Session Border Controller Setting the Wait_Call Activity Parameters 1 Click the Make_Call activity under SIP_Proxy NetTraffic 2 Check the Scenario it contains a set of procedures to originate the call and handle the end call sequence 3 Click the Dial Plan tab and set the Destination Phone Numbers 5557 0000 in this example this sequence should match the one specified as Source Phone Numbers for the SIP_Callee activity this is the criteria for matching the incoming SIP Register messages to this activity 4 Click the Cloud tab and ensure that the Enable SIP Cloud simulation using setting from check box is selected 5 Check whether the dispatching rule in Override default dispatching rules has the proper sequence of phone numbers in the Formula for dispatching field 5551 0000 in this example Setting the Timeline and Objective This configuration contains two Activity Links one for SIP User Agents Caller Callee and one for SIP Registrar and Proxy Server This Activity Links are independent but the test objective for the server should be high enough to accommodate the level of traffic required by the test objective of the Caller Callee Activity Link
124. cutive loss O o Packet loss distribution o Packet loss correlation o Packet errors o Packet duplicates o Calls without media versus calls with media e Audio quality metrics o R Factor PN 915 2611 01 Rev H June 2014 8 Test Case Determining the Maximum Call Setup Rate CPS o MOS o PESQ obsoletes PAMS and PSQM e Video quality metrics o V factor o Absolute MOS V o Relative MOS V o Video loss degradation o Video jitter degradation o Video CODEC degradation The following test methodology provides a structural approach to determining the maximum Call rate supported by a particular DUT e Set performance goals for signaling and media traffic o Targeted calls per second o Use a binary search to determine the maximum call rate Start by using a shorter test trials of 15 30 minutes which can minimize the time required to discover the maximum supported peak rate During the test verify that The call rate is sustained without call failures The SIP QoS is in an acceptable threshold range The RTP QoS is in an acceptable threshold range e Using the measured maximum call setup rate the test should be repeated for a minimum of 72 hours to confirm that the DUT is free of memory leaks and maintains acceptable QoS e After the peak rate is determined short tests can be executed to gather additional data points that can characterize the DUT s QoS The following charts may be used to describe DUT perfor
125. d p31234 gt fEpid activities gt rpid on the phone gt lt rpid other gt online lt rpid other gt rpid activities gt fdm person gt presence gt 4 Create From Template Load From File Edit Options E Change case Modified by user W Case sensitive Delay before execution Static Expression Message body Send audio SDP Offer Send custom message body Edit Custom Send a sip request message Restore Defaults Ok Cancel Figure 235 Configuring custom headers PN 915 2611 01 Rev H June 2014 251 Test Case Telephony Denial of Service 16 After the above steps have been completed link the added activities in the Scenario Editor to enable their execution during the run time Any unlinked procedures will be ignored during traffic emulation by the IxLoad VoIPSIP peer activity Using the cursor link the procedure s OK and Error output to the entry point of the next procedure The last procedure s outputs should be linked to STOP similar to the screenshot below Figure 236 Linking all the configured procedures Setting the VoIPSIP Peer Activity Parameters 17 Click the Dial Plan tab and set the Source Phone Numbers 505 0000 in this example enable the checkbox Overide phone numbers from destination activity and configure the desired destination phone numbers as targets for the TDoS In this example configure 555 0000 18 Click the SIP tab
126. d forged SIP packets to test stability of the device or SUT when these are received Additional test cases can be created by altering the test procedures the order in which they are transmitted or the variables used in the example Please consult the Test Variables section of this test case for more details During the PN 915 2611 01 Rev H June 2014 236 Test Case Telephony Denial of Service test execution it is recommended to monitor the DUT s log files for debug information that might be triggered by the malicious traffic and monitor the system utilized resources If the RAM or CPU load is increasing or system stability is compromised additional verifications should be performed If the device restarts or stops responding properly the test is considered failed and immediate actions should be taken to prevent security failures during live operation Setup In this test topology a single Ixia port emulates the malicious SIP User Agents in a public network initiating traffic towards a SIP Registrar Server as the DUT The SIP User Agents attempt registration with forged content on the DUT an application layer gateway or SBC x es lt x IX FE uf JN 222 Device under Test IxLoad emulated endpoints Figure 222 VoIPSIP topology for Telephony DoS emulation Step by Step Instructions A sample of the configuration as a result of these steps is provided on the Ixia website as DoS scenario crf To import the Compressed Reposi
127. ded 4 Onthe Timeline tab set the Ramp Up Value to 70 5 Set the Ramp Up Interval to 7 second 6 Set the Sustain Time to 10 minutes PN 915 2611 01 Rev H June 2014 165 Test Case Determining the Capacity of a VoIP to PSTN Gateway Assigning Interfaces to the NetTraffics 1 Click Part Assignments from the test configuration panel 2 Add a chassis by clicking Add Chassis 3 Assign a port of one Application Load Module to the SIP NetTraffic In this example the first port of second card 1 2 1 an XMV4 is used for VoIP traffic 4 Assign one port of Adapter Modules for Telephony Boards ADM01 TOB to the PSTN NetTraffic In this example the Card 7 is used for PSTN traffic One Adapter Module for Telephony Boards appears as having a single port indeed one ADMO1 TOB has a single processor that controls all the E1 T1 spans A single adaptor card can accommodate one or two telephony modules each with 2 4 or 8 spans That means a single ADM01 TOB card may have at least 2 E1 T1 spans and at most 16 spans The association of a specific span to a PSTN activity is done in the next step Fort Assignments _ Hide assigned ports F 3 Chassis Chain Assigned Ports Suto refresh status Chassis Chain NetTraffic Total Ports 4ssig Chassis Chain Mew Traffic Flow GM chassis 10 13 10 205 19 67 es Traffic Network 1 of 1 ey Card 1 ACCELERON XP a gt fi Port 1 2 1 Card 2 10 100 1000 LSM xm Ehme Traffice PstnN
128. dia VoIP Calls Test Case Measuring Quality of Experience for Multimedia VoIP Calls Overview In the growing global business environment telepresence has become an important tool for efficient communication between members while assisting time efficiency decrease in travel costs and minimizing carbon emission footprint Telepresence represents a high definition videoconferencing service delivering a virtual face to face meeting experience without the costs and disadvantage of long distance travels The perception is that users across the globe can join a meeting as if they were all in the same room MPLS VPN Metro Ethernet IPSec VPN Headquarters Remote Branch Campus Campus Figure 203 Business to business telepresence application Telepresence from technical aspects represents a combination of technologies to deliver a complex videoconferencing solution The challenge is to preserve a high quality experience for the entire call duration for an almost realistic user interaction By default the teleconferencing solution uses one or more high resolution cameras with additional video channel for presentation content as well as several audio channels originated by all the participating parties Network impairments effect acts differently on the telepresence media components and it s necessary to provide the right level of QoS for the data flows This chapter will not cover the network configuration aspects as it will assist in c
129. dress and transport identifier such as TCP UDP port or ICMP query ID In a traditional NAT sessions are unidirectional being originated only from the private network on outbound sessions Sessions in the opposite direction may be allowed on an exceptional basis using static address maps for pre selected hosts Network Address Translation NAT NAT is acommon method that allows IP addresses to be mapped from one group to another in a transparent way to end users NAT was designed to allow hosts from a private network to use a single device NAT enabled as a gateway to the public network by translating or substituting the private IP addresses of the hosts to the public IP address of the gateway While NAT allows a private network to connect to the public network that is the Internet it also allows a private network to connect to another private network Regardless of how the networks connect the concept is the same and quiet simple to understand Further we will refer to the internal network as private network and to the external network as public network regardless if the external network is the Internet WAN or another LAN We will also refer to the IP addresses from the private network as private IPs and respectively as public IPs for the ones in the public network One of NAT s requirements is to remain transparent to the network that is all devices from the private network are not required to be reconfigured to access
130. dresses which was defined by Internet Assigned Numbers Authority IANA 10 0 0 0 10 255 255 255 172 16 0 0 172 31 255 255 192 168 0 0 192 168 255 255 The three ranges of IP addresses define the private IP addresses that are used within private LANs and not routed to the Internet While the IPv6 addressing scheme will resolve this problem by using a larger address space consisting in 218 addresses its implementation requires the modification of the entire Internet infrastructure For a time IPv4 and IPv6 networks will continue to co exist because IPv4 provides a better solution for small and medium networks IP address translation is useful when a network s internal IP addresses cannot be used outside the network either for privacy reasons or because they are invalid for use outside the network PN 915 2611 01 Rev H June 2014 119 Test Case Using VoIP to Measure NAT PAT Performance In addition a public network topology can change in time for example by the service provider Whenever external topology changes such changes can be hidden from local domain users by centralizing those changes to a single device NAT enabled In this document we will discuss the traditional NAT which consists in a combination of basic NAT and NAPT network address port translation Basic NAT and NAPT are two variations of traditional NAT Basic NAT translation is limited to IP addresses alone whereas NAPT translation is extended to include IP ad
131. e 3 Have any scenario loop failures been reported Check the Loops statistics view Table 47 Statistics highlighting the pass fail result based on call flow execution Statistic Name Questions Are the Successful Loops and Total Loops Total Loops values equal Successful Loops lt 2 Have any Failed Loops Aborted Loops or Failed Loops e Warning Loops been reported Aborted Loops D Note failed aborted and warning loops highlights Warning Loops ee failures at the scenario level PN 915 2611 01 Rev H June 2014 151 Test Case Using VoIP to Measure NAT PAT Performance 4 Has the QoS for signaling met the expected quality Check the Call Times and Delays Statistic views Use the maximum value reported Table 48 Statistics used to determine the QoS for the SIP signaling Statistic Name Value max Questions avg min Is the maximum Call Setup Time less than 4 Call Setup Time seconds End Call Time ne 2 Is the maximum End Call Time less than 2 Talk Time o Media Delay TX RX OO 3 Is the ne eee Delay Tx or Rx less Post Dial Dela S O Men ece 4 Is the maximum Post Dial Delay less than 2 seconds Is the maximum Post Pickup Delay less than Post Pickup Delay 2 seconds For all the stats listed in this table compare their value distribution in time Note Another important factor in establishing the quality of the signaling is the number of retransmissions IxLoad counts those using the SIP
132. e lt S1p 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt s1p 8184443118 myserviceprovider com gt Contact Alice lt sip 8184443118 192 168 1 100 5060 gt Figure 129 Message 1 REGISTER as sent by Alice s phone The REGISTER message will be modified first by the NAT gateway placed at the customer edge This modification will only rewrite the source IP port number in the IP header no modifications are made in the destination IP port or inside the SIP message itself The NAT gateway issues the REGISTER message with the following structure REGISTER sip myserviceprovider com SIP 2 0 Via SIP 2 0 UDP 192 168 1 100 5060 From Alice lt s1p 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt s1p 8184443118 myserviceprovider com gt Contact Alice lt sip 8184443118 192 168 1 100 5060 gt Content Length 0 Figure 130 The structure of message 2 REGISTER after passing the NAT gateway The CE NAT gateway updates its NAT table with the following entry ID SRCIP SRC PORT ae DEST PORT 1 192 168 1 100 75 83 202 16 1025 Figure 131 NAT Table on the CE gateway after REGISTER request is forwarded The REGISTER request next reaches the public interface of the SBC which receives the message on 193 16 148 244 port 5060 The SBC acts as a registrar server for Alice s phone answering the REGISTER request with 200 OK Before transmitting the 200 OK response back to Alice s ph
133. e Phone Numbers field of the Dial Plan Leave the Destination as the Symbolic Link to the Terminate VoIP activity the phone numbers defined there will be used in building the SIP Invite requests Traffici VoIPSipPeer 1 VoIPSip Peer Source IPs The source IP addresses are taken from the associated Network see Traffic Network mapping tables in the test Fhone numbers Phone book entry User defined 1001 1010 gt Ll Use Tel URI parameters onane context example com Phone book Figure 187 PN 915 2611 01 Rev H Destination IPs Traffic _VoIPSipPeer 2 5060 L Override phone numbers from destination activity Phone book entry User defined Verify all settings Dial Plan June 2014 198 Test Case Measuring Quality of Experience for Voice Calls in LTE b Set the SIP Server Address The UE needs to know the IP address of the P CSCF it can be configured as an IP address or as a domain name in the SIP Tab In this case while the P CSCEF is the entry point to the IMS network the IP address of the registrar server is the same The P CSCF knows how to route the registration and call setup requests but for the UEs this is transparent it will send all the SIP requests to the same P CSCF Select the SIP Tab enable Use External Server checkbox set the Server address and Domain name or local IP to the IP address of the P CSCF 22 22 22 1 in this example and enable the Registrar serv
134. e Save the scenario in this example the scenario is save as S P_MakeCall with RTP tst PN 915 2611 01 Rev H June 2014 79 Test Case VoIP Quality of Service in Converged Networks TIP an existing test scenario may be loaded instead of creating it from scratch For example you may load VS_022 DUT_S Pv4 MakeCall EndCall with RTP 33s tst provided with the product Subscriber SIPClient D0 H B coop 4 Gy B X gt o A Ercheck E e Egg B HE i i A Add channel SeeRemove ChE Gompact H Mote zoom 100 workspace x EZALE Editor SIP_MakeCall with RTP Scenario Channel 0 Procedure Librar Others Media SIF H SIP EndCall Initiate Route H SIP EndCall Initiate H SIP EndCall Receive Record Figure 69 SIP Make Call flow 7 In the Servers network add a VoIPSIP Peer activity In this example the activity has been renamed to S PServer Edit the Scenario by adding the SIP Receive Call procedure the Voice Session function and the End Call Receive procedure Save the scenario in this example the scenario is save as S P_ReceiveCall with RTP tst TIP an existing test scenario may be loaded instead of creating it from scratch For example you may load VS_027 DUT_S Pv4 ReceiveCall EndCall with RTP 33s tst provided with the product Trafficl SipServer YoIPSip Peer Scenario Execution bial Plan S Automatic 1 J Cloud Codes ATP J Aud V
135. e 2014 13 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems media delay measures the time after the INVITE message is received and until the first RTP packet is sent vac Us INVITE STAET gt amp STAET A 160 Ay a ca eA Se 200 Media Delay U SSS 5S SS5 S5S Media Delay l ACE eee eieick Stns Scns gt Y i STOP gt coo Tet ETF packet STOF Figure 6 Media delay End call time measures the time to tear down the call When using SIP end call time is measured as the time elapsed between the BYE message and until a 200 OK successful response is received PN 915 2611 01 Rev H TAC BYE START gt fy ll ECT ll Vf STOP 5 Figure 7 End call time June 2014 14 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems e Message retransmissions counts the number of retransmissions required to deliver control plane messages across the network UAC UAS INVITE l s gt E LNVITE l ARR gt E INVITE Retransmissions a X INVITE SSsS gt E INVITE ea ea a area gt 160 ee 200 ee ACK l SSS SS ee gt BYE BYE fe SSeS SSs BYE Retransmissions Aren 200 Fann gt E 200 Retransmissions 200 Seana pe E
136. e DUT will should no longer PN 915 2611 01 Rev H June 2014 244 10 11 Test Case Telephony Denial of Service process any incoming request from that originating source IP to protect the stability of the system and minimize any break in attempts or identity probing After all changes have been made return to the Scenario Editor in the Scenario Channel 0 by pressing the 4P button from the controls ribbon Repeat Steps from 4 to 10 to add a new procedure to the call flow and modify the Expires header value to a number larger than 232 1 The DUT should report this large expire attempts as a security breach attempt or parser error If no validation is in place this case can be considered as failed and proper actions should be performed to minimize the risk during live operations Additionally other alohanumeric characters can be configured to probe the SIP parser engine stability for invalid values PN 915 2611 01 Rev H June 2014 245 Test Case Telephony Denial of Service REGISTER AUTO REQUEST URI SIP 2 0 Via AUTO VIA From AUTO FROM To AUTO TO Call ID AUTO CALL ID AUTO CSEQ Contact AUTO CONTACT Max Forwards T Content Length AUTO CONTENT LENGTH Expires AE AS gan en dA en per ihan hes hs ben hs Create From Template Load From File Edit Options Change case ileal lv Case sensitive Delay before execution Static Expression fo mg Message body Send audio SDF
137. e Files can have the Following characteristics Coding PCM A LAW MU LAW Coding PCM A LA MU Law Sampling frequency 8 kHz Additionally the WAVE Files must be Mono and Sampling Frequency 8 kHz Additionally the WAVE Files must be Mono and Restore Defaults e Restore Defaults aaa mB s Figure 48 Voice Session parameters 6 Save the test scenario flow by clicking Save As located in the Scenario Editor toolbar in a convenient location and with an appropriate name for example H3823_ MakeCall ReceiveCall EndCall with RTP tst he oil Te Ey check A Add Channel EasRemove feGcompact 44 Note Save As button PN 915 2611 01 Rev H June 2014 56 Test Case Determining the Maximum Number of Concurrent Calls 7 Click Full Screen located in the Scenario Editor toolbar Scenario Editor exits full mode 0c H a m M Evcek Fe oe H Badd Channel GarRemove p ECompact 3 Note zoom 100 Full Screen button 8 Save the IxLoad configuration file using File gt Save As save the configuration in a convenient location with a proper name for example IxLoad Voice H323 Concurrent Calls rxf Configuring the Execution Settings for the Make_Call Activity 1 Click the Make_Call activity 2 Click the Execution page 3 Set the script to be executed a single time during the Sustain Time by setting Run for to A number of loops and entering 7 4 Select the Grace
138. e activity are exposed through TCL API the parameters in Scenario Editor are not accessible from TCL API To make the configuration using the parameters from activity 1 From Network1 click the H 323 peer activity labeled Make_Call 2 The configuration page shows Scenario Editor PN 915 2611 01 Rev H June 2014 54 Test Case Determining the Maximum Number of Concurrent Calls Click Full Screen located on the Scenario Editor toolbar Scenario Editor appears in full screen mode HE B Add Channel Ea Remove ECompact Note zoom 100 coe x T Ey check Sa as a a 78 HE Full Screen button Locate the Voice Session script objects used by Scenario Editor The scenario has two functions one on the caller side and one on the callee side workspace l Scenario Editor YH_002_B2B_H323v74_NC_Basic_Call_with_RTFP e Scenario Channel 0 Activity Make_Call Traffic Trafficl Network1 Column Originate Link VolPLink41 H323 Library wp H323 Make Call ih H323 Receive Call E H323 End Call H323 RAS Library Media Library ATP Generate DTMF ATP Detect DTMF ATP Generate MF ATP Detect MF ATP Generate Tone ATP Wait For Tone FTP Talk ATP Listen ATP Voice Session ATP Path Confirmation ATP RTP Control ATP Multimedia Session Figure 47 Scenario Editor in full screen mode see highlighted Talk script objects PN 915 2611 01 Rev H June 2014 55 Test Case Determining the Maximum N
139. e length CAVLC This is a less demanding data processing compared to CABAC but more complex in terms of coefficient coding e Various loss resilience features as Network Abstract Layer Flexible Macroblock Ordering or Frame Numbering to better compensate the transmission s channels information losses or errors The codec has advanced techniques of encoding and decoding high definition media with a better bit rate efficiency There still was the need for a more scalable method to deliver the information when endpoint capabilities that demand the same stream are different Annex G of H 264 MPEG 4 AVC was an answer that allows the encoding of the main streams with one or more subset bitstreams The main advantage of H 264 Scalable Video Coding SVC is that it uses less bandwidth as it drops specific portions of data from the original stream and displays at reasonable quality the resulted video signal It delivers the video to the user s display with a drop in spatial resolution decrease the virtual screen size decrease in temporal resolution use a lower number as frame per second or lower quality video signal thus consuming a smaller bandwidth These scalars can be used independently or in combinations for better converge of the media delivery over impaired network infrastructures H 264 SVC can use the base layers for a video stream and construct delivery over multiple enhancement layers for media transmission It is essential for teleconf
140. e some applications VoIP is one of those applications Various workarounds exist but those add an extra layer of complexity and processing which can result in a performance decrease The central part of all the NAT operations is the NAT table which typically resides in the memory of the device implementing NAT The NAT table has a dynamic behavior it is populated as new connections are created and after the connections are closed the associated bindings are removed Larger NAT table allow more bindings can be tracked PN 915 2611 01 Rev H June 2014 123 Test Case Using VoIP to Measure NAT PAT Performance The basic NAT can be further classified in two types e Static NAT e Dynamic NAT Static NAT As described earlier traditional NAT sessions are unidirectional originating from the private network while sessions in the opposite direction may be allowed on an exceptional basis using Static address maps for pre selected hosts Most NAT devices today allow the network administrator to configure permanent entries into the NAT table with the goal of allowing inbound sessions to reach designated devices within the private network for example a Web server This type of configuration is referred to as static NAT or port forwarding 172 16 1 2 16 7 Port 80 5 209 132 176 30 22 80 HTTP Server Figure 111 Static NAT for INCOMING connection PN 915 2611 01 Rev H June 2014 124 Test Case Using VoIP to Measure N
141. e stress test is ready for execution The test objective will be configured to 700 Loops per Second 21 Click Timeline amp Objective from the test configuration panel 22 For the VolPSIPPeer1 set the test Objective Type to Loops Initiated per Second 23 Set the test Objective Value to 700 PN 915 2611 01 Rev H June 2014 203 Test Case Telephony Denial of Service 24 Set the Sustain Time to 1 hour Execute the Test Start Map the port according to the test environment and run the test by clicking the button Test Variables Table 88 Test configuration Parameter Name Current Value Additional Options P Type Pv6 Type of traffic SIP sianalin Audio Video T38 Audio Video RTCP yp g g traffic Other SIP methods User defined ee Number of loops per second User defined a Call duration User defined Mix with data protocols for Not included Any combination of data protocols example FTP HTTP Telnet supported by IxLoad successfully calls canceled calls Mix of call flows unanswered calls busy calls call forward call transferred call Mix of call features hold retrieve Results Analysis The DUT shall be monitored for the following e Memory size memory allocation de allocation issues while o Phone numbers are added deleted to from the user table o Invalid SIP Translations are continuously detected and stability faults are not detected e CPU usage e Size of LOG files PN 915 2611 01 Rev H Ju
142. eSubscribers network add an IPTV Video Client activity In this example the activity has been renamed to PTVClient Drag and drop the lollipop of the PTVClient activity to PTVServer activity Automatically the command Play is added in the Command List with the parameters shown in the table below Table 31 IPTV Client Play command parameters Parameter Value Server IP Address Traffic1_IPTVServer 554 Meda Steam SSS Leave the rest of the parameters at their default values 5 Inthe HomeSubscribers network add an HTTP Client activity In this example the activity is renamed to HI TPClient Drag and drop the lollipop of the H7 TPClient activity to HT TPServer activity Automatically the command Get is added in the Command List Set the Page Object to 728k html The parameters of the GET functions are shown in the table below Table 32 HTTP Client Get Parameters Parameter VELT Destination IP or IP Port Traffici_HTTPServer 80 Page Object 128k html Add a Think command and set the Random Duration Between 1000 and 3000 ms This will reduce the throughput to around 22Mbps for 40 users the test objective Leave the remainder of the parameters at their default values 6 In the HomeSubscribers network add a VoIPSIP Peer activity In this example the activity has been renamed to S PClient Edit the Scenario by adding the SIP Make Call Complete procedure the Voice Session function and the End Call Initiate procedur
143. e_Call Phone numbers 4lias Override phone numbers from destination activity C Phone book entry User defined 818501 1000 ser definer FOpoOoOoooooC Type of Alias Type of number National number Numbering plan SDN telephony numbering Fhone book Verify all settings Restore defaults Figure 50 Make _ Call Dial Plan configuration example for Network 1 Configuring the H 323 Settings for the Make_Call Activity in Network 1 1 Click the H 323 configuration page of Make_Call 2 Verify that the Enable signaling on this activity check box is selected 3 Verify that the Enable FastStart check box is cleared 4 Verify that the Enable Tunneling check box is cleared 5 Set Q 931 User User to 878501 1000 6 Set Q 931 Display to xia 818501 1000 PN 915 2611 01 Rev H June 2014 58 Test Case Determining the Maximum Number of Concurrent Calls 7 Verify that the Enable RAS check box is cleared Traffici Make Call VoIPH323 Peer Dial Pan F323 Terminal capabi e signaling on this activity iF unchecked all H323 script Functions will be SKIPPED H323 Specific Settings Advanced Signalling Options Bandwidth 4nd Terminal Type _ Enable FastStart Bandwidth fin Kbps o4 _ Enable Tunneling Terminal Type Terminal Entity without Mc 50 Enable Parallel H245 Versions Send Call Alerting H 22o Wersi 0 0 8 2250 0 5 te _ Send Call Proceeding ae H 245 version 0 0 8 245 0 9
144. eated channel to an activity appears Eroro Execution ial Pan SIP automatic 75 Cloud Codecs RTP aud ve Fax 7 36 Fax 30 RTP other Dg W coup amp Gs B x 7o A icheck E e e e e pin gt B add Channel Soe Remoye eel Gompact H Moke Foor 100 Create Configure Linked Activity in Activity Link YoiceLink1 Workspace This activity Traffic Networkl 51PMakecall uses scenario channel 0 from a best scenario C Program Link VoiceLink1 Filesi Ixia Load Repository Mihai BB July 2010 51P PSTN basic call tst with 2 scenario channels H323 Library H323 RAS Library Media Library ATP Generate DTMF Step 1 Select a column in the IxLoad Flow Editor ATP Detect DTMF RTP Generate MF Step 2 Select a NetTraffic in the selected Column FTP Debeck MF Step 3 Be ee ve Enter the name For a new activity vorsipPeerd 7 ATP Wait For Tone oan ATP Talk Select the protocol lvorsp o YS ATP Listen Do you want to create anew oIP activity or configure an existing one to use scenario channel ss 1 EE Parent activity ATP Voice Session ATP Path Confirmation ATP RTP Control ATP Multimedia Session E T38 Fax Session Figure 153 Add anew test scenario channel and map it to an activity 4 Inthe Create Configure Linked Activity in Activity Link VoiceLink1 window click Terminate at steo1 Traffic2 PstnNetwork1 at step 2 and Selec
145. eceive_Call Activity 1 Click the Audio tab 2 Ensure that you have selected the Enable audio on this activity check box 3 Specify the clip to be played 4 Specify the duration of the play this will determine the call duration 5 Select the MOS and One Way Delay check boxes if these metrics are of interest for the test you want to configure Enable audio on this activity if unchecked all audio script Functions will be SKIPPED Play Settings _ Enable jitter buffer Format PCM Duration 32785 ms Size 524556 bytes Output level 20 dBm 7 C Play for clip duration or TalkTime all objectives except Channels me C Play For Perform Qow Type OF Service ips TOS DSCP Calculate One Way Delay _ Generate silence Figure 37 Audio Settings PN 915 2611 01 Rev H June 2014 35 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring the Timeline and Objective 1 Select Timeline amp Objective from the test configuration panel 2 Set the test Objective Type to Calls Initiated per Second 3 Set the test Objective Value to 700 4 Onthe Timeline tab set the Ramp Up Value to 700 5 Set the Ramp Up Interval to 7 second 6 Set the Sustain Time to 5 minutes 7 Set the Ramp Down Time to 40 seconds Network Traffic Mapping Objective Type Objective Value Iteration Time Total T iz 3 Mew Traffic Flow G M Activity Links Eh volPLinki Calls Initiated Per Second Timeline 1 0000 05 45
146. ect a specific logical component of the DUT testing with media enabled can help discover additional issues which may not be detected without media An example is testing a device that implements a proxy element for SIP only signal forwarding Other capacity tests measure the maximum number of registered users that can be concurrently maintained The overhead added by registration or other keep alive messages exchanged between servers and clients is frequently overlooked when testing For example a SIP registration that requires authorization includes four messages versus nine messages for a call session which also requires authentication On systems that need to handle thousands of users the overhead is quite significant due the fact that online users will re register every hour or less On some systems the registration can recur as often as every 60 seconds This overhead is important for systems that implement registrar and proxy components on the same server In general the DUT needs to be monitored during testing for the following e Memory leaks e Stability e Ability to sustain performance for an extended period e Signaling quality of service e Media quality of service Testing Challenges in VoIP Networks Convergence Due to voice traffic s high sensitivity to delays packet loss and jitter a converged network can have a negative impact on the voice quality when is not properly configured Quality of service QoS configur
147. ed by the session initiation protocol SIP video and HTTP servers PN 915 2611 01 Rev H June 2014 76 Test Case VoIP Quality of Service in Converged Networks Step by Step Instructions The final IxLoad configuration as a result of these steps is provided on blackbook ixiacom com Web site see IxLoad 5 10 Voice QoV in Converged Networks crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test 1 Configure a test with two traffic flows one for subscriber simulation and one for servers simulation Use the network parameters defined in the following tables Table 29 Network parameters for the Subscriber simulation A ANETE Type of Traffic Flow Element Increment 0 0 0 1 Gateway 0 0 0 0 Gateway Increment 0 0 0 0 MSS RX 1460 Table 30 Network parameters for the Servers simulation Network Name Servers Type of Traffic Flow Element Increment 0 0 0 1 Gateway 0 0 0 0 Gateway Increment MSS RX PN 915 2611 01 Rev H June 2014 77 Test Case VoIP Quality of Service in Converged Networks This example assigns a single IP address to all servers HTTP IPTV and SIP For high scale tests it is possible to use multiple application load module ports to simulate servers in
148. een For multipoint telepresence calls the Service Server sitting between the endpoints delivers additional delay due to internal queue processing or load processing IxLoad assists in measuring the required time for multimedia channels to reach from one end to the other as part of QoS assessment when measuring the MOS score The One way delay metrics measures the time needed from one codec s output to other s codec input with a high precision for a multitude of current active channels A set of tools and techniques are required to preserve high quality for real time high definition video and audio over converged IP network This minimizes the effect of impairments such as bandwidth rate policies latency jitter and packet loss By design voice and real time video should be granted high priority service while other non critical applications may have a lower priority enforced Several questions are to be asked before actual telepresence service deployment which can assist in building the converged network service prioritization requirements e ls telepresence the only traffic objective e Which additional real time media services will be deployed over the converged network e Which critical applications will share the same network resources PN 915 2611 01 Rev H June 2014 221 Test Case Measuring Quality of Experience for Multimedia VoIP Calls e Are there other types of traffic identified that did not match those above What are those
149. eeseeeteueesseetaeess 235 COMAC al een te ene eee eee ee eee ee eee ee eee 257 PN 915 2611 01 Rev H June 2014 V Voice over IP How to Read this Book The book is structured as several standalone sections that discuss test methodologies by type Every section starts by introducing the reader to relevant information from a technology and testing perspective Each test case has the following organization structure Overview Provides background information specific to the test case Objective Describes the goal of the test Setup An illustration of the test configuration highlighting the test ports simulated elements and other details Step by Step Instructions Detailed configuration procedures using Ixia test equipment and applications Test Variables A summary of the key test parameters that affect the test s performance and scale These can be modified to construct other tests Results Analysis Provides the background useful for test result analysis explaining the metrics and providing examples of expected results Troubleshooting and Provides guidance on how to troubleshoot common Diagnostics issues Conclusions Summarizes the result of the test Typographic Conventions In this document the following conventions are used to indicate items that are selected or typed by you e Bold items are those that you select or click on It is also used to indicate text found on the current GUI screen e Italicized items are
150. eleration check box to allow 8 000 RTP streams per Acceleron XP port Configuring the Dial Plan for the Receive_Call Activity 1 Click the Dial Plan tab 2 Set the Source Phone numbers by clicking Specify and using the sequence 979501 0001 3 For Destination IPs click None no calls are generated from this activity with the test scenario used Traffic Receive_Call VoIPSip Peer Source Destination IFs The source IP addresses are taken From the associated IFs None Network see Traffic Network mapping tables in the test Phone numbers Override phone numbers From destination activity C Phone book entry User defined 919501 0001 User defined FOfooo00000 C Use Tel URI parameters Figure 32 Dial Plan settings for Receive_Call activity PN 915 2611 01 Rev H June 2014 32 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring the SIP Parameters for the Receive_Call Activity 1 Click the SIP tab 2 set the SIP Port number to 5060 3 For back to back tests verify that the Use External Server check box is cleared Traffice Receive _Call VoIPSip Peer Enable signaling on this activity SIP Port s060 iF unchecked all SIP script Functions will be SKIPPED Transport settings Use external server Maximum message size on UDP 1024 _ Override transport specified in scenario Server port _ TCP send immediate Outbound proxy Enable
151. en an error occurs on VoIP an error is logged in the Even Viewer window indicating the endpoint ID the error type and description and hints to resolve the error Analyzer and Traffic Packet Viewer you can enable traffic capture per port to save memory the RTP outbound packets are not captured to minimize the size of the capture you can apply filters using the tcodump syntax Test Variables Table 75 Parameter Name Current Additional Options EU Up to 8 000 concurrent active endpoints in calls with audio streams can be emulated by a single 1G port of an Xcellon Ultra NP card To increase the number of channels you have to allocate enough resources in terms of IP Addresses and Phone Numbers User Names see section 16 Set the Timeline and Objective The configuration is created for capacity testing Other test objectives are available for testing the rate of call setup Concurrent Calls supported by the access network the CPS test objective can be used Beside the value of Call Per Second Rate you have to specify the number of emulated endpoints or the call duration the talk time these three parameters are correlated Test Objective Channels CPS Number_of_Endpoints Call_Duration where Call_ Duration Talk_Time Call_ Setup Time Overhead_ Time To increase the CPS you have to use more endpoints or reduce the Talk Time PN 915 2611 01 Rev H June 2014 212 Test Case Measuring Quality of Experience for Multime
152. ents and CODECs Bandwidth is directly affected by the phone s encoder and the expected number of simultaneously active calls A higher number of active conversations calls corresponds to higher bandwidth requirements Audio CODECs which implement speech compression can differ significantly in bandwidth requirements speech quality encoding delay loss resiliency and computational requirements Hence the selection of a CODEC can directly affect the number of active calls that a link can support and the voice quality In general reducing the nominal bandwidth that is payload throughput degrades speech quality Because the relationship between bandwidth and speech quality is not a linear one some CODECs can significantly reduce the nominal bandwidth required while preserving a good quality For example G 729 can reduce the nominal bandwidth 8 times compared to G 711 while maintaining good speech quality Many CODECs support silence suppression mechanisms which can reduce the required bandwidth up to 50 percent Effect of Delay on Conversation It is well Known that network delay leads to two way conversation difficulties This effect can be better described using an example Let us assume that Alice and Bob have a phone conversation using an IP network with a high round trip delay for example 500 ms Say that Alice decides to interrupt Bob while he is talking Due to the delay Alice will continue to hear old information from Bob
153. equence of Phone Numbers in the VolPSIPPeer2 dial plan e Increase the number of virtual IP Addresses for VolPSipPeer2 activity under Cloud tab Set the Sustain Time to 5 min 17 Ports mapping Map ports to the TrafficNetworks You will need a pair of ports connected to the EPC system the DUT PN 915 2611 01 Rev H June 2014 209 Test Case Measuring Quality of Experience for Voice Calls in LTE Running the test 18 Save the configuration Save he configuration using the Save button in Quick Access Toll Bar or the option Save under File or the keys Ctrl S If it first time you save the configuration you will be prompted to enter a name for the rxf configuration and then you ll be prompted to enter a name for the tst file The tst file contains the SIP call flow 19 Run the test Results Analysis The EPC SIP and RTP stats need to be analyzed in this configuration EGTP Data Rates kbps All Ports EGTP Packet Rates All Ports gt Figure 200 EPC Packet Rates PN 915 2611 01 Rev H June 2014 210 Test Case Measuring Quality of Experience for Voice Calls in LTE The Packet Rate increases when new calls are established but majority of the packets are RTP Calls ValFSip Px Figure 201 VolP Calls The number of Attempted Connected calls on the Originating side and the number of Received Answered calls on the Terminating side must be equal in a successful test Any difference means the calls
154. er Activity Parameters 1 Click the Caller activity under SIP_Caller NetTraffic 2 Click the Scenario tab and check whether the call flow is the required one In this example the call flow is Register Make Call Play Audio Streams and Initiate end call Refer the Black Book sections describing how to create and change the call flow in the Scenario Editor or the IxLoad VolP User Guide for instructions of how to modify the call flow Click the Dial Plan tab and set the Source Phone Numbers 5551 0000 in this example Set the Destination IP address to the IP address of the SIP Proxy server 40 40 50 1 5060 in this example Set the Destination Phone Numbers 5552 0000 in this example Click the SIP tab and set the Server Address to the IP address of the SIP Proxy server or the IP address of the DUT private side depending on the type of DUT 7 Click the Audio tab and ensure that the Enable audio on this activity check box is selected this activity will play and receive audio on the established calls PN 915 2611 01 Rev H June 2014 178 Test Case Determining the Performance of a Session Border Controller Setting the SIP_Caller Activity Parameters 1 Click the Callee activity under SIP_Callee NetT raffic 2 Click the Scenario tab and check whether the call flow is the required one In this example the call flow is Register Receive Call Play Audio Streams and Wait End call Refer the Black Book sections describing how to cre
155. er checkbox Traffic VoIPSipPeer 1 VoIPSip Peer Dial Gn SIP Amat Enable signaling on this activity if unchecked all SIP script functions will be SKIPPED Transport settings Maximum message size on UDP 1024 _ Override transport specified in scenario _ TCP send immediate Enable FQDN resolution Authentication UAC Username user0 001 010 Password 12345 AKA authentication settings Select configuration lt None gt Edit configurations L Type Of Service TOS DSCP Figure 188 PN 915 2611 01 Rev H C Registrar server SIP Port 5060 Use external server VoLTE settings Server address Enable Call Control of dedicated bearer Server port 5060 Domain name or local IP L Outbound proxy _ Auto register simulated user agents m Override registrar Construction of SIP messages L Override default contact settings L Override default destination domain name or host port _ Use Tel URI scheme for Source _ Use Tel URI scheme for Destination Transfer address Verify all settings Restore defaults Set the SIP Server IP address June 2014 199 Test Case Measuring Quality of Experience for Voice Calls in LTE c Set the Authentication credentials In the real deployments the SIP requests for registration or call setup are authenticated by the server The UE has to use the proper credentials to have access to the service Several
156. er to an IP phone via an SBC and CE NAT gateway SRC 192 168 1 100 5060 SRC 172 16 100 9 5060 thie eal ee ae ae DST 193 16 148 244 5060 SRC 75 83 202 16 5060 DST 172 1 61 00 2 5060 aaa 1 REGISTER DST 193 16 148 244 5060 Alice i 3 REGISTER 192 168 1 100 3 2 REGISTER ee onies Enterprise NAT NAPT i ontrolle 0 a Service Provider s ar i IN Core Network 192 168 1 1 uuu 193 16 148 244 4 200 OK 6 200 OK 75 83 202 16 5060 l Carol DST 192 168 1 100 5060 SRC 193 16 148 244 5060 Media Gateway 192 168 1 101 DST 75 83 202 16 1025 172 16 100 3 SS A A G SRO 72 16 1002 5060 SRC 192 168 1 1 5060 1025 5 200 OK DST 172 16 700 9 5060 Figure 132 Path of the 200 OK response initiated by SBC When the 3 REGISTER request is sent by the SBC to the registrar server it will have different Call IDs and transaction information compared to the initial request SRCIP_ SRCPORT____ DESTIP_ DESTPORT _172 16 100 9 5060 172 16 100 2 5060 __ REGISTER sip myserviceprovider com SIP 2 0 Via SIP 2 0 UDP 172 16 100 9 5060 From Alice lt sip 8184443118 myserviceprovider com gt tag 5000c5000 To Alice lt sip 8184443118 myserviceprovider com gt Contact Alice lt sip 8184443118 172 16 100 9 5060 gt Content Length 0 Figure 133 Structure of message 3 REGISTER Assuming no NAT device is between SBC and registrar server the response
157. er1 Traffic Traffic3 Public Column Originate Link VoIPLink2 SIP wait Mes 100 01 200 01 Timeout Error SIP 200 Ok 4 IP100 Trying 43 K OK Transport Failure E i SIP wait Req Timeout Error Figure 98 Register test scenario Channel 0 sends the registration request and then waits for the response If the response is 401 Unauthorized then a new registration request is sent this time with following authorization header REGISTER AUTO REQUEST URI SIP 2 0 Via AUTO VIA From AUTO FROM To AUTO TO Call ID AUTO CALL ID CSeq JAUTO CSEQ Contact AUTO CONTACT Max Forwards 70 Content Length AUTO CONTENT LENGTH Authorization AUTO AUTHORIZATION PN 915 2611 01 Rev H June 2014 110 Test Case Determining the Maximum Transaction Rate for VoIP Protocols The call flow on the second channel emulates the SIP registrar with Authorization enabled It waits for a Registration request and responds with 100 Trying and 401 Unauthorized t then waits for a new Registration request with an Authorization header and responds with 100 Trying and finally 200 OK Using this call flow every emulated endpoint will exhibit the following traffic Flow Summary p 6 message s and 2 endpoint s in 4 Fria pay SIP 20 1 1 1 5060 SIP 20 1 100 1 5060 More SIP Endpoint SIP Registrar 1 00 00 30 957907 REGISTER sip 20 1 100 1 SIP 2 0 2 00 00 30 960122 SIP 2 0 100 T
158. erencing especially over loss prone and low bandwidth networks like the Internet The advantage of H 264 SVC is that one encoder can send a single media stream to a multitude of heterogeneous endpoints where independent decoding is performed depending on the technical capabilities Several independent tests have shown that in certain situations when H 264 SVC codec is used and the packet loss that a network can introduce is between 20 and 40 the user can have a decent intelligible image compared to the packet loss acceptance criteria of H 264 AVC where the loss tolerances is up to 5 PN 915 2611 01 Rev H June 2014 220 Test Case Measuring Quality of Experience for Multimedia VoIP Calls One of the noticeable disadvantages of using H 264 SVC is inter operability issues One vendor for example can implement specific optimization or algorithms for low bandwidth video delivery that cannot deliver the same results on the device of another vendor The inter operability requires large amount of tests to ensure that most of the cases are covered and the video delivery to the endpoint converges to a good quality For most telepresence applications the video cameras capture and encode a static image with little movement so the overall video rate can drop up to 35 of the per stream average bitrates Dealing with variable codec such as H 264 the challenge comes when the peak rates exceed the average estimations This generates a significant traffic o
159. erminal Type _ Enable FastStart Bandwidth fin Kbps 64 _ Enable Tunneling Terminal Type Terminal Entity without ME 50 Enable Parallel H245 Versions Send Call Alerting H 225 Versi 0 0 8 2250 0 5 tt Send Call Proceeding ciel H 245 Version 0 0 8 245 0 9 Registration Admission and Status RAS _ Enable RAS Retry and Timeouts For RAS Messages Maximum Retry Count Timeout fin secs Gatekeeper O 931 User Lser 918501 1000 _ Hexadecimal Byte Stream Display Ixia 918501 1000 Figure 57 Configuration example for H 323 settings of Receive_Call activity H 323 Normal Start Configuring the CODEC Settings Page for the Receive_Call Activity The second group of H 323 phones is configured to use G 711 uLaw Packet time 20 ms 160 bytes per frame Traffic Receive_Call VoIPH323 Peer Audio Codecs Settings Payload type mens ilaca Packet kine 20 ms 160 bytes per Frame Incoming payload type T 6711 Law Outgoing payload type Oo 2 Iarll Alaw i a re ITU T 5 711 is a standard to represent amp bit WARNING The outgoing payload type number compressed pulse code modulation PCM samples For will be used For RTP header payload type when signals of voice Frequencies sampled at the rate of sending packets and viceversa 5000 samples second Please note that the outgoing payload type of a G 711 encoder creates a 64 Kbps bitstream sender must be equal to the incoming payload type of a recei
160. ermine the transaction rate for Registration and Instant Messaging Registration Rate In a traditional wired network each device has a fixed location that the switch determines from its associated circuit In VoIP networks a phone s location is not pre determined a subscriber can move his soft IP phone and plug it in anywhere in an IP network It is necessary for each VoIP device to identify itself to the network and the network must grant access to the device SIP registration is performed in one or two transactions depending on whether the registration server is configured to require authentication or not Bob s SIP phone SIP Server REGISTER 200 OK Figure 90 SIP register without authentication Bob s SIP phone SIP Server REGISTER 401 Unauthorized REGISTER 200 OK Figure 91 SIP register with authentication PN 915 2611 01 Rev H June 2014 106 Test Case Determining the Maximum Transaction Rate for VoIP Protocols After an endpoint has registered the network has the IP address location of the endpoint In the case of SIP the Registrar server holds that information Using the known location of an endpoint SIP proxies will know how to route a connection Even after a successful registration an endpoint may become unavailable without the Registrar server or SIP proxies becoming aware of the event To protect against this the Registrar server may ask the endpoint to periodically retransmit its registration method If
161. erride transport specified in scenario _ TCP send immediate Enable FODN resolution seconds ae DNS expiration timeout _ Cache FODN resolution Authentication UAC User name Password AKA authentication settings Select configuration lt None gt Edit configurations Figure 237 Dial Plan SIP Automatic TLS Cloud Codecs RTP Audio Video Fax T 38 Fax T 30 SRTP SIP Port 5060 Use external server Server address Server port Domain name or local IP Outbound proxy Registrar server _ Auto register simulated user agents _ Override registrar Construction of SIP messages _ Override default contact settings Override default destination domain name or host port Domain name or Host Port Use Tel URI scheme for Source _ Use Tel URI scheme for Destination Use automatic headers SIP tab configuration example 20 Check whether the dispatching rule in Override default dispatching rules has the proper sequence of phone numbers in the Formula for dispatching field 5557 0000 in this example Setting the Timeline and Objective In this example the DUT will receive all the malformed traffic configured in the test Depending on the verbosity level of the logs a large amount of information might be logged It is recommended to start with lower traffic levels and increase after the initial debug has been performed and th
162. ers Undispatched Messages under SIP Cloud view PN 915 2611 01 Rev H June 2014 184 Test Case Determining the Performance of a Session Border Controller Dispatched Messages VoIPSipCloud WE Undispatched Messages VolPSipCloud Sent Messages YoIPSipCloud Figure 171 Dispatching SIP Messages Stats 5 Has the QoS for signaling met the expected quality Check the Call Times and Delays Statistic views Use the maximum value reported Table 68 Statistics used to determine the QoS for the SIP signaling Statistic Name Questions Is the maximum Call Setup Time less than 4 seconds ls the maximum End Call Time less than 2 seconds _ Is the maximum Media Delay Tx or Rx less than 4 seconds Is the maximum Post Dial Delay less than 2 seconds Is the maximum Post Pickup Delay less than 2 seconds For all the stats listed in this table compare their value distribution in time 6 In some cases the DUT is not involved in the media path In other cases for example when the SIP User agents are in different networks the DUT handles the media as well In this case the following quality of media indicators should be checked RTP MOS RTP QoS RTP Advanced QoS RTP Jitter Distribution RTP Consecutive Lost Datagram Distribution and RTP Streams statistic views Table 69 MOS statistics Statistic Name Questions How do the last values reported by the RTP MOS Best and RTP MOS Best RTP MOS Worst compare with each
163. ess and replace it with the host address 192 168 1 101 The same lookup operation retrieves the destination port as well which remains unchanged Telnet Client 192 168 1 102 2500 receives an incoming connection with the response from the external Telnet Server 207 46 197 32 23 ee a re ae re ere ee ee ne ee eee ee l Private Network Internal Public Network External Il NAT NAPT 192 168 1 101 24 DST IP 192 168 1 101 DST PORT 2 500 DSTIP 75 83 202 16 A T5 83 202 16 22 pdt SRC IP 207 46 197 32 SRC PORT 80 SRC IP 207 46 197 32 SRC PORT 23 192 168 1 1 24 Figure 117 Incoming connection Telnet response This example demonstrated how two hosts can share a public IP address in the same time Now let s see how NAPT works when two hosts initiates the same type of connection using same source port and also uses the same destination IP address and port In fact the outbound destination and port will not change the behavior of the NAPT enabled router but two connections sharing the same source port number affects the behavior of the router As we described in the previous examples the port numbers are preserved The next figure illustrates how the outgoing sessions are handled when they use the same port For both connections NAT changes the source address to its public address 75 85 202 16 For the first connection the port number is not changed because no other connection exists with port 32 000 How
164. et structure for the 200 OK response The SIP message includes several header fields that include IP and port information related to the delivery of the message e The top Via header indicates the IP address and port number of the last hop of the SIP message The responses must follow the path indicated by Via header e The Contact header indicates the current location of the endpoint This location can be further used by the remote party to send subsequent messages such as ACK and BYE unless a Record Route mechanism is forcing the messages to always pass the PROXY PN 915 2611 01 Rev H June 2014 133 Test Case Using VoIP to Measure NAT PAT Performance e The session description protocol SDP is used to negotiate how the media session must be handled The following two fields carry information related to the media IP address and port number that can cause NAT problems o The connection c field includes IP addresses used to send and received media o The media m field includes the port number used to send and receive media Even assuming a successful establishment of the SIP session the media path will fail because the c and m fields included in the SDP body of the INVITE request sent by Alice s phone will use the private IP address 192 168 1 100 and a port number dynamically selected by the phone for media transmission and reception Hence after the call is established the incoming media from the public network will be SRC
165. etworkt E Port 1 2 1 amp E Port 1 7 1 AB Port 1 2 2 i Port 1 2 3 LB Port 1 2 4 ey Card 4 Voice Quality Resource Module OLI Bort Lt On Card 6 4DMO1 TOB E1 T1 08 TOB E1 T1 04 3 jE Port 1 6 1 Running For 9 Hr 11 Min HE Card 7 ADMO1 TOB E1 T1 04 44 i hell Port 1 7 1 On Card 9 A4CCELERON NP Port 1 9 1 Port 1 9 2 Port 1 9 3 Port 1 9 4 Fort 1 9 5 Dwkiag Show IP Assignments Assign PSTN Interfaces Figure 156 Assigning interfaces to VoIP and PSTN NettTraffics PN 915 2611 01 Rev H June 2014 166 Test Case Determining the Capacity of a VoIP to PSTN Gateway Assigning PSTN Interfaces 1 Inthe Port Assignment page click Assign PSTN Interfaces 2 Select the desired span in the Available Ports pane and associate it with the PSTN Network Range In this example Span s is used it PSTN Interface fissignment gt The panel on the left shows the all assigned ports and their physical interfaces available for assignment to the net traffic ranges The panel on the right shows the PSTN net traffics created in the test and For each net traffic the ranges and their assigned interfaces Available Ports PSTN Network Ranges EHE Chassis 10 1 10 205 19 87 Chia Traffic2 PstriNetworkl ih BE Card 7 40M01 TOB E1 T1 04 EF ty amp PSTNDigitalRange 1 an HuntGroupl 1 HEE Port 1 7 1 Span 2 of Board 1 7 1 1 e Board 1 CLE Mode Standalone Ref Available Digital Spans 3 4ssigned Digital Spans 1 m
166. ever because the second connection is initiated also from the same port 32 000 NAT changes the port number to the first application port available which is different than the active port numbers on the public interface In our example the source port number is changed to 1025 PN 915 2611 01 Rev H June 2014 128 Test Case Using VoIP to Measure NAT PAT Performance HTTP Client 192 168 1 100 32 000 initiates an outgoing connection to an external HTTP Server 209 132 176 30 80 Private Network Internal I Public Network External il NAT NAPT 192 168 1 100 24 SRC IP 192 168 1 100 SRC PORT 32 000 SRC IP 75 83 202 16 SRC PORT 32 000 75 83 202 16 22 DST IP 209 132 176 30 DST PORT 80 192 168 1 1 24 i a a i Private Network Internal Public Network External Il NAT NAPT 192 168 1 102 24 SRC IP 192 168 1 102 SRC PORT 32 000 SRC IP 75 83 202 16 SRC PORT 1025 DST IP 209 132 176 30 DST PORT 80 192 168 1 1 24 T ae DST PORT 80 Figure 118 NAPT example when two hosts initiate an external connection using the same port Port numbers between 1 and 1023 are reserved for well known service port numbers Applications port numbers start with 1024 and end with port number 65535 Full Cone NAT Restricted Cone NAT Port Restricted Cone NAT and Symmetric NAT Based on how NAT devices handle the UDP traffic we can differentiate the following NA
167. ey are different and if the measured PESQ score Is 5 percent or more less than the expected score then the source of degradation will be determined by analyzing other statistics provided by IxLoad These include MOS packet loss jitter and delay Shown below are the results of three test runs the first is without any degradation the second with packet loss and the third with transcoding errors IxLoad provides a set of voice clips that have been selected to cover all the characteristics of the human speech PESQ scores vary with the content and level of the voice clip and more importantly with the codec used A table of scores for various clips obtained by connecting load module ports to each other in a back to back fashion with volume levels and codec is provided in the IxLoad documentation If the DUT performs ideally then the scores for PESQ should match the values listed in this table In the current example the test is configured to use the voice clip FemaleMale_Mix3 with an Active level of 20dBm The DUT transcodes the clip from G 711uLaw to G 729A The expected score Is 3 512 as marked in the excerpt from the PESQ LE scores table shown in Table 4 below Table 40 Baseline of PESQ LE scores Codec Active G711 G723 iLBC Level Mulaw 5 3kbps 13 33 G729A G729B kbps 25dBm 4 441 3493 3739 3 578 3289 Female1_Seq1 35dBm 4399 3 437 3683 3 611 3601 20dBm 4 457 3383 3512 3512 3 663 25dBm
168. fic subset of calls for example the calls ona trunk or on a specific VLAN Traffici MakecCall VoIPSip Peer Enable audio on this activity iF unchecked all audio script Functions will be SKIPPED Play Settings Enable jitter buffer Clip US5_042 way Buffer size 100 ms Format M A Use compensation Output level 20 dBm E nol ms Play For clip duration or TalkTime all objectives except Channels Max dropped consecutive packets _ Play For Perform Qov Type OF Service Units of Channels TOSIDSCP Class 1 0x20 value 150 Channel Selection Evenly Spaced Channels Perform MOS Calculate One Way Delay _ Generate silence Figure 79 RTP parameters at activity level for PESQ test 5 Select the Perform MOS check box This setting specifies that QOV MOS will be computed for all calls The MOS scores in conjunction with PESQ scores will allow identification of the cause of any quality of voice degradation 6 Select the Enable jitter buffer check box Keep the default value for Jitter Buffer Size if the expected jitter in the network under test is known to be less than 20 ms or set it to the appropriate value 7 Set the test objective and timeline in accordance with the DUT s capacity The number of concurrent calls and the CPS rate may affect the QoV through the DUT PN 915 2611 01 Rev H June 2014 94 Test Case Subjective Quality of Voice 8 Assign the ports that will generate traffic and assign t
169. following standard metrics can help assessing the QoS for signaling protocols Call setup time measures the time required to setup a call including the call acknowledgement This metric is significant if the called party answers immediately In practice someone may answer after a variable time Post dial delay measures the time required to receive the first ring back notification after the last digit of the destination phone number was dialed The post dial delay is significant for the caller A large post dial delay is perceived as a call that did not go through because the caller does not receive any indication of the call progress in the ring back tone A post dial delay value of less than four seconds is recommended for IP telephony systems When post dial delay exceeds four seconds the caller may hang up and attempt a new call Post pickup delay measures the time after the call was answered until the first media packet is received The post pickup delay has a direct impact on the user experience Normally a caller starts to be affected with delays higher than four seconds Media delay measures the time to receive the first media packet after a call setup message was sent While the call may be connected via a delayed conversational path media delay can still be annoying to the caller PN 915 2611 01 Rev H June 2014 7 Test Case Determining the Maximum Call Setup Rate CPS e End call time measures the time
170. for Destination Transfer address TOS DScCP Verify all settings Restore defaults Figure 189 Authentication Credentials PN 915 2611 01 Rev H June 2014 200 Test Case Measuring Quality of Experience for Voice Calls in LTE d Set the Audio Codec to AMR WB VoLTE requires AMR codec for the speech communication This is a multi rate codec optimized for speech with capability to adapt to variations of network conditions IxLoad supports both AMR NB and WB versions Select the Codecs tab in the table of audio codecs select the first one and choose AMR WB codec from the drop down list You have the option to change the order of preferred codecs The SDP will be automatically built with the parameters configured in this page Trafficl VoIPSipPeer 1 VoIPSip Peer Audio Codecs Spe Codec mode 0 6 60 kbps Incoming payload type 39 am PA V Payload format Octet aligned format Outgoing payload type 99 Bandwidth efficient format i i WARNING The outgoing payload type number aiei Eme M me i butes per frame will be used for RTF header payload type when iiS we ye par a sending packets and viceversa p Please note that the outgoing payload type of a _ Enable VAD sender must be equal to the incoming payload AMR WEB Adaptive Multi Rate WideBand is a multi mode type of a receiver codec that supports 9 speech encoding modes with bit rate between 6 60 and 23 85 Kbps 16000 Hz sampling frequency 50 fps
171. format T1 1 544 Mbps divided in 24 time PN 915 2611 01 Rev H June 2014 155 Test Case Determining the Capacity of a VoIP to PSTN Gateway slots is used in North America The 24 time slots of a T1 link or 32 of the E1 can be used to transfer signaling data or both of them Channel Associated Signaling CAS and Integrated Services Digital Network ISDN are two of the standards defining how the signaling and media is sent over the TDM links Setup The example below shows the step to originate 30 concurrent calls from SIP and terminate them on an E1 span set to ISDN media will be sent in both directions on the established calls The example uses one port of an Application Load Module for example Acceleron XP and one E1 T1 port of one of the Telephony Boards 944 0002 TOB E1 T1 02 944 0003 TOB E1 T 1 04 or 944 0004 TOB E1 T1 08 The test setup topology is shown in the next figure PSTN Network 1 F j EJ IXIA SIP endpoints PSTN endpoints Figure 146 SIP to PSTN test setup Note This configuration requires a real gateway the test configuration cannot be executed in back to back by connecting Acceleron and TOB interfaces together The gateway is configured to route the SIP calls received on the IP address 192 168 1 100 port 5060 to its E1 interface PN 915 2611 01 Rev H June 2014 156 Test Case Determining the Capacity of a VoIP to PSTN Gateway Step by Step Instructions The final IxLoad configuration
172. ful Ramp down check box this is the default setting Note This option forces the users to hang up the call when the ramp down request is received in the middle of the call Trafficl Make Call VoIPH323 Peer Fun For Loop delays D The entire test duration Before 1st loop T i Between loops j ms Phone Number Mapping Rules for H3z3 Graceful Rarmpdown Phone Number Use consecutive values per acti Graceful Rarmp down Verify all settings Restore defaults Figure 49 Configuration example for Execution Settings page of VolPPeer1 5 Set the Phone Number Mapping Rules for H 323 a Phone Number Use consecutive values per activity Configuring the Dial Plan for the Make_Call Activity in Network 1 1 Click the Dial Plan configuration page of Make_Call 2 Set the Source Phone numbers by selecting the User defined check box 3 Set the source phone number using the sequence 878501 1000 4 Keep the Type of Alias Type of number and the Numbering plan options to their defaults PN 915 2611 01 Rev H June 2014 57 Test Case Determining the Maximum Number of Concurrent Calls 5 Set the destination to Traffic2_Receive_Call by clicking the corresponding symbolic link from the list available for destination IPs Traffici Make Call VoIPH323 Peer Source Destination The source IF addresses are taken From the associated IPs j j Network see Traffic Network mapping tables in the test IPs Traffice_Receiv
173. g steady state check the TCP failures High reached for HTTP and IPTV TCP timeout and RST packets can indicate that the device is unable to traffic Throughput goes up handle the load and down Not all SIP attempted calls If the device incurs UDP packet loss or the routing delay of packets is are connected high some SIP calls are not completed Change the transport protocol used for SIP from UDP to TCP or keep the transport set to UDP and enable retransmissions for the SIP activities MOS score is less than 4 41 The expected MOS score when using the G 711 uLaw codec is 4 41 If the score is lower be sure this is not the effect of the codec if the test is configured to use a different codec the score will be lower Verify the One Way Delay statistics a value bigger than 150 ms affects the MOS score Verify packet loss a packet loss of 3 percent reduces the MOS score by more than 0 5 SIP calls are not established Verify that the DUT is routing SIP messages properly If the DUT has SIP at all intelligence it may include a proxy server that requires registration of the user agents before executing the tests If this is the case modify the test scenario to include a register procedure before originating calls PN 915 2611 01 Rev H June 2014 86 Test Case Subjective Quality of Voice Test Case Subjective Quality of Voice Overview Speech quality in a telephony system is a subjective judgment that depends on technical at
174. gnaling traffic from one endpoint to the other Verify the Event viewer for error information No media traffic is flowing To ensure that media traffic is flowing through the system verify the firewall rules to allow RTP and RTCP to pass through If necessary enable Analyzer function and capture the SIP conversation Analyze the SDP information exchanged for additional details on the sockets negotiated for media exchange Measured MOS has low To overcome this verify in the IxLoad statistics the delay jitter and packet quality rating loss introduced by the SUT Depending on the observations adjust the configuration to allow a better connectivity between the emulated endpoints Conclusions This test has offered the methodology to measure the QoS for media delivery in the case of business to business telepresence applications The SUT configuration and performance are determined for an optimum user experience using the existing configuration template PN 915 2611 01 Rev H June 2014 233 Test Case Telephony Denial of Service Test Case Telephony Denial of Service Overview As a general use the term denial of service represents the attempt of sending malicious traffic towards a destination until the available resources are no longer available for the users of interest or the recipient of the traffic In telephony this concept has a very simple example the destination phone number is called over and over again by an autom
175. gt Figure 8 SIP message retransmission A large number of retransmissions received or transmitted by a DUT can decrease the overall DUT performance and cause instability The number of messages retransmitted must be maintained below 5 percent in a deployed IP telephony system and should be kept close to zero when testing a DUT in isolation e Call establishment ratio measures the ability of a called party to successfully connect a call and establish a conversation Gail nenien Rabo e a a Git CSLADLLSAILETLL ee Y of Calls Attempted ro e Call completion ratio measures the ability of a called party to successfully connect the call and complete the call by initiating or receiving the appropriate disconnect request of End Calls Completed Cen ee ees a PN 915 2611 01 Rev H June 2014 15 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Objective The objective of this test is to attempt a steady call rate of 100 calls per second cps while applying a constraint of 8 000 simulated users The test will also highlight how to apply a constraint of call hold time or a dual constraint of simulated users and call hold time The instructions provided for this test describe how to configure SIP activities to simulate SIP IP phones SIP trunks and SIP networks with users behind them They will also helping the reader to understand how Ixia s IxLoad traffic generator can be configured to achieve a con
176. h a bandwidth request 12 times higher the registration rate requires high performance from the network and VolP devices especially SBCs and registrar servers Message Rate In addition to call control SIP defines a mechanism to support instant messaging It consists of a two SIP message exchange the originator sends a MESSAGE method followed by a response from the receiver PN 915 2611 01 Rev H June 2014 107 Test Case Determining the Maximum Transaction Rate for VoIP Protocols Bob s SIP phone SIP Server MESSAGE 200 OK Figure 93 Instant messaging transaction In spite of its simplicity the proliferation of instant messaging has placed a high load on networks and VoIP devices especially SBCs and SIP proxies Objective This test attempts to establish a steady 500 registrations per second The test demonstrates how to use the transaction rate test objective to measure the registration or instant messaging rate The following instructions explain how to configuring IxLoad to emulate SIP endpoints and SIP Registrar server Setup A pair of Acceleron ports is used to simulate 8000 user endpoints in a medium sized Enterprise deployment that uses a Registrar server in the public network The test runs for 5 minutes Enterprise NAT NAPT REGISTRAR amp PROXY Figure 94 Test Topology IxLoad generates and receives the traffic passing the NAT IXIA IXIA Step by Step Instructions The final IxLoad
177. he Enable video on this activity check box 4 Select the Clip to be played from the list 5 Set the duration of the call to 3 minutes using the Play for parameter PN 915 2611 01 Rev H June 2014 148 7 8 Test Case Using VolP to Measure NAT PAT Performance To enable calculation of video MOS V MOS relative V MOS absolute and V Factor select the Perform MOS check box to also automatically select the Calculate One Way Delay check box Traffici Make Call YoIFSip Peer Scenario Execution Dil Plan SIP J Automatic 15 loud Codecs RT audio Video J Fac 38 Fax 30 5RTP ther Enable video on this activity if unchecked the Multimedia Session script Function will be SKIPPED Fire_ avec mp4 Codec H264 Duration 12800 ms Size 2012176 bytes Bitrate 1225 kbps Ignore Hint Track Play For clip duration or TalkTime all objectives except Channels Minutes Conference mode Type OF Service TOS DSCP Figure 144 Video settings Note if the test objective will remain Calls Initiated per Second at step 3 select the Play for Talk Time option to allow the test objective to calculate and pass the call duration to the script Select the second activity Receive_Call and repeat steps 2 to 6 Save the repository file by using the File Save CTRL S command Configuring the Timeline and Objective 1 2 Click Timeline amp Objective from the Test Configurat
178. he Quality of voice Resource Module If PESQ is enabled for at least one activity at least one Quality of voice Resource Module must be assigned The resources of this module are shared by all activities that have PESQ enabled Port Assignments FP amp Chassis Chain Assigned Ports Chassis Chain ar a Tii PAarric ECMwOor a GE Card 2 10 100 1000 ALM T8 1GB on Trafficl Networkt 2 of 2 Port 1 2 5 ff Port 1 2 1 fi Port 1 2 6 Po ef Port 1 2 3 Port 1 2 7 a heh Traffic Network 2 of 2 fi Port 1 2 8 fi Port 1 2 2 g BE Port 1 2 4 Figure 80 QoV resource assignment for a PESQ test 9 Start the test and activate the following statistic views RTP MOS QoV PESQ RTP QoS and RTP Per Channel Each of these views is provided on a per protocol basis For example if two VoIP signaling protocols are used in same test for example SIP and H 323 two RTP MOS views are available one for simulated SIP User Agents and one for H 323 simulated endpoints In this test only one signaling protocol is used SIP so only one RTP MOS view will be shown Test Variables Depending on the characteristics of the DUT the test should be repeated with the following variations PN 915 2611 01 Rev H June 2014 95 Test Case Subjective Quality of Voice Test Tool Variables Table 38 Test tool variables PESQ test Parameter Name Current Value Additional Options Codec Ptime 20ms 10 30 ms 10 30ms 3 min 60 min To extend
179. he SIP INVITE message 192 168 1 100 5060 Hence the SBC will return the 200 OK SDP back to the public address and port used by the CE NAT to send 2 INVITE 75 83 202 16 1025 rather than using the private IP address and port number 192 168 1 100 5060 used in the Via header of the 2 INVITE message Upon receipt of the 2 INVITE the SBE SBC component initiates a new call to the SIP proxy server located in the service provider network which connects the call to a PSTN network not displayed in the figure It initiates a new call by sending 3 INVITE SDP which establishes a separate call with the proxy server In this message the IP header the SIP headers and the SDP body will use the signaling IP port address 172 16 100 9 5060 while the following SDP lines are used for the media IP address and port c IN IP4 172 16 100 10 m 25000 audio RTP AVP 0 18 101 REGISTRAR amp PROXY SRC 172 16 100 9 5060 172 16 100 2 5060 DST 172 16 100 2 5060 3 INVITE SDP Session Border Controller Service Provider s Core Network 193 16 148 244 4 200 OK SDP 5060 i 17216 100 2 5060 DST 172 16 Media Gateway 172 16 100 3 INVITE SDP c IN IP4 172 16 100 10 m audio 25000 200 OK SDP c IN IP4 172 16 100 3 m audio 30000 172 16 100 10 172 16 100 3 25000 30000 RTP Figure 138 Establishment of media between an SBC and Media Gateway PN 915 2611 01 Rev H June 2014 143 Test Case
180. he caller may hang up and attempt a new call Ac erin ain See Weais DAs INVITE S TART gt gt fy 302 ACK SRD gt INVITE Tm nanan cham aaa la a gt Ve 150 STOP gt lt Figure 4 Post dial delay measured on a redirected call e Post pickup delay measures the time from when the call was answered until the first media packet is received In SIP this corresponds to the time elapsed from when the 200 OK response for the INVITE was sent until the first RTP packet is received UAC UAS INVITE isa ie acca a gt 180 le ee ele a ee es 200 lt lt START ACK oy aa ae gt PPL i ist RTP packet gt lt TOP Figure 5 Post pickup delay The post pickup delay has a direct impact on the user experience Normally a caller starts to be affected with delays higher than four seconds e Media delay measures the time to receive the first media packet after the call setup message was sent This measurement can be applied for both the calling party and the called party When using SIP the media delay for the calling party is calculated as the time between the INVITE message and the first RTP packet is received For the called party the PN 915 2611 01 Rev H Jun
181. he information concerning the calls that are affected is provided in the RTP Per Channel view E RTP Fer Channel VoIPSip i ii H el t EB lbh TH Stat Name 4 J 110 200 1354 170 Cardd2 Port02 vVolPSipPeer Channelooo 4 110 200 154 170 Card02 Port02 vVolPsipPeer Channel0oo 5 110 200 154 170 Cardd2 Portd2 volPsipPeer2 Channel00026 110 200 134 170 Card02 Port VolPSipPeer2 Channelooo27 110 200 134 170 Card02 Port2volPsipPeer2Channeldooz8 110 200 154 170 Card0e Port02 vVolPsipPeer Channeloo0 9 110 200 134 170 Card02 Porto2 VolPSipPeer Channel00030 110 200 134 170 Card02 Porto2 VolPSipPeer2 Channel0oo31 110 200 134 170 Card02 Port02 vVolPSipPeer Channel0o032 110 200 134 170 Card02 Port02VolPSipPeer2 ChannelOo033 110 200 134 170 Cardd2 Portde VoIP sipPeers Channelooos4 110 200 134 170 Card02 Porto2 vVolPSipPeer Channel00035 Aan OOD DA Te dA AD Tie ne a dT h wee Slice Figure 85 PN 915 2611 01 Rev H 2 Oo A 8 A A A 8 8 A 8 A m June 2014 44 616 44 616 44 616 44 616 44 616 47 900 43 824 43 384 43 516 43 296 43 516 AT Cic 55e Bytes Recei Packets Rec RTP per channel statistics in case of Packet loss ost Packet 100 Test Case Subjective Quality of Voice Transcoding errors lf there are no degradations in the packet transmission then the MOS score will be perfect but the PESQ score can still be lower than
182. he location and the name of the TST file for example SIP_MakeCall ReceiveCall EndCall with RTP tst hs Te Ey check A add Channel EG Remave ee Compact 4 Note Save As button 7 Click Full Screen in the Scenario Editor toolbar Scenario Editor exits the full screen mode e H m M Evcek Fe oe D HE B dd Channel EorRemove ECompact 3 Note oom 100 Full Screen button 8 Save the IxLoad repository file as an RXF file using the File gt Save As menu option change the location and the name of the RXF file for example IxLoad Voice SIP Call Setup Rate rxf Configuring Execution Settings for the Make_Call Activity 1 Select the Make_Call activity 2 Click the Execution Settings tab PN 915 2611 01 Rev H June 2014 25 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 3 Set the script to continuously repeat during the Sustain Time the Sustain Time is the test duration to be set later in this example using the Run For the entire test duration option 4 Select the Graceful Ramp down check box this is the default setting Note If the ramp down event in the timeline occurs while the phone is in the middle of the call the Graceful Ramp Down option will stop the RTP traffic and will continue executing the next actions by skipping any media action and executing only signaling actions In this way the signaling flow will end by sending
183. hting the pass fail result based on call flow execution Statistic Name Questions 1 Are the Successful Loops and Total Loops values equal Successful Loops 2 Have any Failed Loops Aborted Loops or Total Loops Failed Loops Warning Loops been reported Aborted Loops oe Note failed aborted and warning loops highlights Warning Loops 7 failures at the scenario level PN 915 2611 01 Rev H June 2014 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 4 Has the QoS for signaling met the expected quality Check the Call Times and Delays Statistic views Use the maximum value reported Table 5 Statistics used to determine the QoS for the SIP signaling Statistic Name Vailue max Questions avg min h i Il Time han 4 Call Setup Time Is the maximum Call Setup Time less than End Call Time Talk Time Media Delay TX RX seconds Is the maximum End Call Time less than 2 seconds Is the maximum Media Delay Tx or Rx seconds Is the maximum Post Pickup Delay less than 2 seconds For all the stats listed in this table compare their value distribution in time Post Pickup Delay Post Dial Del less than 4 seconds a Zz ls the maximum Post Dial Delay less than 2 Note Another important factor in establishing the quality of the signaling is the number of retransmissions IxLoad counts those using the SIP Retransmitted Msgs statistic located under the SIP Messages
184. ice Session Properties OO Talk Parameters Listen Parameters Advanced Settings Output Settings Listen Parameters Advanced Settings Output Settings Delay Before Execution Listen Settings Static Expression f Listen duration 23000 ms Z 0 m f Wee Talk Time for all objectives except Channels Talk Parameters Perform Gov measurements Clip a Output level on JEm Output Volume 20 dBm U 0 7745 A 600 ohm ak 05042 wav T Output level 20 dBm PCM 8000 Hz Mono Size 524556 bytes Time 32 85 ms Play i timels f Repeat continuous for fico sec f Use Talk Time for all objectives except Channels f Use Global settings Plays one or more WOVE Files and listens for audio RTP at the same time The 5 Plays one or more WAYE Files and listens for audio RTP at the same time The r wave Files can have the Following characteristics wave Files can have the Following characteristics Coding PCM A LAW MU Law Coding PCM A LAW MU Law Sampling frequency 8 kHz Additionally the WAVE files must be Mono and Sampling Frequency 8 kHz Additionally the WAVE Files must be Mono and Restore Defaults FE Restore Defaults ae E E Figure 22 Voice Session configuration example for the Calls Attempted per Second objective 6 Save the test scenario flow as a TST file by clicking Save As available in the Scenario Editor toolbar change t
185. ice with a time lag There is also a strong reverberation effect that cause severe degradation of speech quality making it difficult to understand the speaker or even hauling created by feedback Advanced video processing algorithms For high resolution delivery with features such as flicker removal hue and chroma compensation or image alteration reduction due to improper lighting or optical focus Professional teleconferencing solutions have a multitude of elements that work in synergy to deliver high quality output that allow a comfortable and realistic human interaction Telepresence Call Flow There are several layers involved to establish a telepresence call as there are multiple ways to deliver video conferencing to a single or multiple participants The typical layers involved in establishing a call are as follows 1 User Interface Layer This requires user interaction with the device to program the required action This can be a graphical application running on the device or computer a voice responsive control or a push of a specific pattern on dial pad This is schedule set up or places the call Account policy is there to reinforce the user access rights to various actions that are allowed to be performed for the profile This is the primary point of access and it also delivers various technical details to the other layers involved to establish the call Conference Control Layer This is mainly responsible for resource alloca
186. ideo J Fax 7 38 Fax 7 30 5RTP Other Ce Hal mcr tS OB xXlo o lA aor e eee S 8 ip LE Be Add Channel Ss Remove PE Gompact fal Note Zoom 100 workspace 1 scenario Editor SIP_ReceiveCall with RTP SIP MakeRegistration First L i e Scenario Channel 0 H SIP MakeRegistration SIP ReceiveCall Busy Here H SIP ReceiveCall No Answer gt 3 oe D OK Or SIP ReceiveCall RecordRout eE OEE mi F Error Pay Error A E a H SIP ReceiveCall PT TaREReRegistr ation au F H SIP Send Instant Message H SIP UnHold Initiate SIP Recei FTF oice Ses UT SIP EndC Figure 70 SIP Receive Call test scenario PN 915 2611 01 Rev H June 2014 80 Test Case VoIP Quality of Service in Converged Networks 8 Set the following parameters for the SIPClient activity part of the HomeSubscribers network Table 2 Configuration parameters of the SIP simulated UAs VoIPSIP activity Configuration Parameter Value Page Dial Plan Destination IP A by clicking from Automatic Enable retransmissions Enableaudio Enable Enable imer butter Enable PerformMOS rable Leave the remainder of the parameters at their default values 9 Set the following parameters for the SlIPServer activity part of the Servers network Table 33 Configuration parameters of the SIP simulated trunk VolPSIPServer activity Configuration Parameter
187. ideo MOS The Video Quality Assessment Engine measures individual media streams delivering ratings based on gathering specific information such as e Type of frame and Group of Pictures structure e Per frame video quality e End to end roundtrip and system delay e RTCP metrics e Average and maximum bitrate per frame type e Content descriptors such as level of detail motion panning frozen or blank video e Network specific quality metrics and much more Objective This test methodology helps evaluate the quality of transport of the inter campus transport network The simulated application is the videoconferencing between the headquarters and a remote branch over the deployed MPLS network or the METRO Ethernet link The user experience rating is measured for the audio and video streams played using the MOS rating for the business to business telepresence deployment model PN 915 2611 01 Rev H June 2014 223 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Setup The telepresence endpoints are emulated by IxLoad The transport network between the endpoints represents the SUT target for QoS qualification The appropriate configuration must be deployed prior to test execution to allow networking access from one side to the other gt lt Tr as Uxa Emulated Emulated Endpoint Endpoint System Under Test Figure 208 Environment for telepresence application testing NOTE IxLoad client should be installed
188. igure 234 Configuring SDP offer in the originating request PN 915 2611 01 Rev H June 2014 200 Test Case Telephony Denial of Service 15 Repeat steps from 4 to 10 to add a new procedure to the call flow and add a large number of custom defined headers The reason is to validate the DUT capacity to properly parse the received messages and not compromise the security or stability When the message size is large the expected performance should slightly decrease and no critical errors should occur If the custom headers defined are recognized by the DUT these should be properly processed regardless of the place where these are found in the message AIP Send Request Properties Params Behavior Flow Manager Extract Variables Authentication Output Settings REGISTER AUTO REQUEST URI SIP 2 0 Wia AUTO VIA From AUTO FROM To AUTO TO Call ID AUTO CALL ID IcSeq AUTO CSEQ Contact AUTO CONTACT IMax Forwards 70 Content Length AUTO CONTENT LENGTH Expires 3600 X S5esgion 5tatus Open Client Type SWEndpoint K Client VYersion 1 2 3 Media 5upport Limited ontent type application pidf xml cpresence xmlns rpid urn ietf params xml ns pidf rpid xmlins dm urn ietf params xmil ns pidf data model zmlns urn ietf params xml ns pidf entity pres ixia ixiacom com tuple id 12345 gt status gt basic gt open lt basic gt status gt tuple gt Om person i
189. imedia VoIP Calls Ports Chassis Chain Assigned Ports G New Traffic Flow Che Traffici Networki Non Aggr 1 Port 1 1 3 Port 1 1 4 Port 1 1 5 Port 1 1 6 Port 1 1 7 Port 1 1 8 Port 1 1 9 Port 1 1 10 Port 1 1 12 ae Port 1 1 11 ee ee ee eee eee eee Figure 220 Assigning ports to NetTraffics 16 Start the test execution by triggering the Start Test gt from the ribbon commands At this time the SUT configuration should have been adjusted accordingly to the scope of test As necessary modify QoS policy and firewall rules to allow SIP signaling to pass from one endpoint to another as well as to the RTP and media signaling data Results Analysis The following questions provide guidelines on how to interpret the results during test execution and how to identify issues that can arise during the objective s measurement Have any call failures been reported Check the Calls VoIPSip view Table 78 Reported statistics Statistic Name Value Questions Calls Attempted as Calls Connected Have any call attempts failed Calls Received Compare Calls Answered Calls Attempted and Calls Received with End Calls Received Calls Attempted and Calls Connected End Calls Completed PN 915 2611 01 Rev H June 2014 230 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Has the test objective been achieved Check the Channels VoIPSip view Table 79 Statistic Name
190. inding When the last session based on an address or address TCP UDP port tuple binding is terminated the binding itself may be terminated PN 915 2611 01 Rev H June 2014 130 Test Case Using VoIP to Measure NAT PAT Performance NAT Traversal for VoIP The number of broadband users has increased significantly in the last years leading to a faster adoption of the VoIP services by both home users and enterprises Regardless of the endpoint type that is soft phone or hard phone or service type residential or business the majority of the VoIP endpoints use private IP addresses that are mapped to public IP addresses using NAT NAPT a common function of the broadband access routers Additionally VolP endpoints may be situated behind one or more firewalls The following characteristics of the VoIP protocols impose challenges while traveling through NAT 1 A VoIP call can be seen as two separate sessions which needs to be correlated a A signaling session that uses protocols such as SIP H323 MGCP and H248 that Establishes and tears down a media connection Negotiate the common set of capabilities Agree on the source and destination IP address and port s b A media session which represents the actual conversation that happens between the IP addresses and media ports dynamically negotiated by the signaling session 2 A VoIP session uses the source and destination IP address and port numbers inside the IP payload
191. interface or dial a known extension according to the endpoint characteristics One endpoint sends a SIP INVITE to the other endpoint with details about the required media for transmission The headers defined in PN 915 2611 01 Rev H June 2014 215 Test Case Measuring Quality of Experience for Multimedia VoIP Calls SDP contain data for media negotiation such as codecs in use codec attributes and required bandwidth Depending on the capabilities of the remote party the call determines if it is to continue or new SDP information is to be negotiated After the initial contact is established a second negotiation takes place to enable the real time video negotiation and establish the data paths required Both endpoints negotiate as a set of common codec that consider the media capabilities such as video resolution video bitrate and supported list of codecs Both endpoints next start to present the media information received from the other side The telepresence call is now active To preserve a feedback mechanism during the call vendors can implement methods like SIP SUBSCRIBE SIP OPTIONS or SIP NOTIFY as update or keep alive mechanisms The negotiation details are thus enforced by the device capabilities to encode and decode specific formats as well as functionality constraints within the configuration A multipoint teleconference flow is similar to the business to business call model in that one of the peer endpoints is the telepresence Se
192. ion panel Set the test Objective Type to Channels Set the test Objective Value to the number of concurrent calls desired Under Timeline set the Ramp Up Value Set the Ramp Up Interval to 7 second Set the Sustain Time to cover the test duration desired Note The Sustain Time must be higher than the Talk Time It is also good practice to add some extra time for example 30 seconds as a buffer for any delays that may occur during the call setup and end call phases Hence sustain time should have a value higher than 3 min 30 sec 3 30 min PN 915 2611 01 Rev H June 2014 149 Test Case Using VoIP to Measure NAT PAT Performance Test Variables Test Tool Variables Table 43 Parameter Name Current Value Additional Options Type of traffic Audio Video T38 Audio Video RTCP 1t01 1t0N Nto 1 and NtoN IP Mapping rules for signaling and media Number of users avaliable Useren OOOO Number ofactve cals Useren O O o GallRate Useren S CaiDuaion Usera OOOO Audio Video CODEC type packet G 711u G 711 G 729 G 723 G 726 iLBC size amp frequency H264 AMR Mix with data protocols for Not included Any combination of data protocols example FTP HTTP Telnet supported by IxLoad VolP Protocol used SIP H323 SKINNY MGCP H248 successfully calls canceled calls Mix of call flows unanswered calls busy calls call forward call transferred call Mix of call features i hold retrieve DUT Test Variables
193. is larger than the expected message length the SIP engine generates a buffer overflow taking down the entire services for all the serving users This is a critical issue as the amount of traffic required to take down the service is extremely low while the effect on the services is severe This malformed signaling traffic methodology for TDoS attacks will be treated in this test case and it will cover a few scenarios that can be achieved with IxLoad as a traffic initiator The RFC4475 contains a large suite of test cases for SIP torture testing this document does not intend to cover the entire suite and it addresses a limited set of examples The scope is to demonstrate the flexibility of the IxLoad test suite and should be always used in a lab environment for testing purposes Typically when a malformed packet is received by the DUT this should be dropped and no longer processed Processing invalid information or parameters with invalid values might lead to memory leaks buffer overflows or in extreme situations may compromise the application s Capability to process future requests A more severe case can be when the application allows remote access to the system resources after such manipulation of malicious traffic gt g lt gt he gt Z lt gt Z 5 TDosS Target Initiators Users Figure 221 Generic topology for targeted Telephony DoS Objective This test demonstrates the steps required to configure IxLoad to sen
194. itter distribution statistics Statistic Name VEIO Questions a a Jitter up to 1 ms rll Jitter up to 3 ms See Jitter up to 5 ms ace Jitter up to 10 ms rel Jitter up to 20 ms 1 Assuming Jitter was reported what is the distribution of the Delay Variation Jitter values Jitter up to 40 ms Jitter over 40 ms Table 9 Distribution of RTP Consecutive Lost Packets Statistic Name Value s Questions Packet Sequence Three Packet Sequences Ten Packet Sequences or More Packet Sequence C We L EE an 1 Assuming that packet loss was reported ga me T To m what is the distribution of the lost RTP or Five Packet Sequences packets PN 915 2611 01 Rev H June 2014 42 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Statistic Name Concurrent RTP Streams Concurrent RTP Streams max Table 10 RTP Streams WEI G Questions Assuming that packet loss was reported what is the distribution of the lost RTP incoming RTP packets Are any calls with RTP wit 1 Does this number match the number of incoming RTP packets Calls Connected 2 Troubleshooting and Diagnostics The following table summarizes some of the common issues that may be encountered when running a call rate test The attempted call rate is not sustained for the entire test duration Calls are attempted at a constant call rate only from time to time followed by intervals without any calls attempted The
195. ive the first ring back notification after the last digit of the destination phone number was dialed Post pickup delay measures the time between when the call is answered until the first media packet is received Media delay measures the time to receive the first media packet after the call setup message was sent End call time measures the time to tear down a call Message retransmissions a count of the number of retransmissions under ideal conditions that is no packet loss or delay PN 915 2611 01 Rev H June 2014 47 Test Case Determining the Maximum Number of Concurrent Calls When the DUT simultaneously handles signaling and media for example SBCs it is important that media traffic be enabled for all calls In these test cases you must check that media traffic meets QoS requirements by inspecting the following measurements e Assessing the transport of media O O O O Packet loss jitter and one way delay Packets sent versus packets received Inbound outbound throughput Maximum consecutive loss packet loss distribution packet loss correlation Packet errors packet duplicates Calls without media e Audio quality metrics O O O R Factor MOS PESQ obsoletes PAMS amp PSQM e Video quality metrics when applicable O O V Factor Absolute MOS V Relative MOS V Video loss degradation Video jitter degradation Video CODEC degradation PN 915 2611 01 Rev H June 2014
196. k on the Add Net Traffic button in the Toolbar or right click and select Add Net Traffic Networks and Traffic TrafficFlow1 w t o EEI f Zoom Out FP _ Add NetTraffic db Add Subscriber P Add DUT P Add PSTN NetTraffic dp Add Column 3e Remove 4 Move Left Figure 180 Add Net Traffic by right click on the Network and Traffic pane PN 915 2611 01 Rev H June 2014 194 Test Case Measuring Quality of Experience for Voice Calls in LTE 7 Set the IP address of the emulated P CSCF One P CSCF server is emulated in this configuration Set its IP Address to 22 22 22 1 and the Count to 1 Networks and Traffic TrafficFlow1 v Terminate a B Ethernet 8 Unconfigured IPv4 16 0 0 0 1 Figure 181 IP Address of the emualted P CSCF Add the Emulated P CSCF 8 Add the SIP Cloud activity Move the mouse over the Traffic element a plus sign will be shown click on it and select the VoIPSIP Cloud activity Bulk MGCP Bulk SIP H248 MGC H248 MGW MGCP CA MGCP GW VoIPH323 te Figure 182 Add a SIP Cloud activity this is the emulated P CSCF PN 915 2611 01 Rev H June 2014 195 Test Case Measuring Quality of Experience for Voice Calls in LTE 9 Set the Distribution Group of the IP address range to IP Round Robin The SIP Cloud requires that the distribution of the network ranges by port to be set to IP Round Robin Select the Traffic element i
197. kets Endpoints connected in calls without receiving any media are counted using the Calls without RTP packets Statistic All calls were connected but the Verify that requests to disconnect the calls are not received expected maximum throughput is not from the DUT during the Talk Time reached The incoming throughput and outgoing Verify the Codec Distribution and RTP Packet Distribution throughput values are lower or higher Statistics these may be used to confirm that the calls than expected negotiated the desired audio video CODEC Problems may be fixed by correcting the DUT configuration or by setting the same CODEC on both calling and receiving activities PN 915 2611 01 Rev H June 2014 73 Test Case Determining the Maximum Number of Concurrent Calls Test Variables Test Tool Variables Table 26 Parameter Name Current Value Additional Options IPversin gt IPversin gt IPv4 IPv4 si IPE Up to 96 000 per card assumes AAO ARAS ee ASM1000XMV12 RTP cards Call Duration User defined Network Configuration Static IPs IP DHCP IP IPsec IP PPPoE Codec parameters for the negotiated audio UTON bpf every G 711 G 729 G 723 G 726 i LBC AMR CODEC type packet size amp frequency 20ms RTCP H 323 Call Setup Mode Slow Start Fast Start Tunneling IP Mapping rules for signaling and media 1to 1 1toN Nto 1 and N to N Mix with data protocols for example FTP Not included Any combination of data protocols HTTP Telne
198. l Timeouts ISDN cas voice wE Library Settings and Outputs VoIP T1 E1 Layer 1 bearer channel ma Functia utput Results Bearer channel capability BEAR_CAP_SPEECH 08 Scenario Editor Defaults od Application Workspace Destination number type INTERNATIONAL r Destination number plan ISDN Origination number type INTERNATIONAL ka Origination number plan ISON x Reject call reason NO ROUTE TO TRANSIT NE Initiate RESTART procedure while shutting down spans Restore Defaults Save Library Settings Load Library Settings Cancel Figure 160 Global Settings for Voice Activities 3 Start the test by clicking Run Results Analysis The following questions provide guidelines on how to recognize specific problems during or at the end of the test execution 1 Has the test objective been achieved Check the Calls view PN 915 2611 01 Rev H June 2014 169 Test Case Determining the Capacity of a VoIP to PSTN Gateway 1 Is the number of Attempted calls equal with the numbers of Connected calls Are all the originated SIP Attempted Calls 300 PAlicucsnnesied SIP Connected Calls Same PSTN Received Calls value for PSTN Answered Calls all stats 2 Is the number of Attempted calls equal with the number of received and Answered calls Note If the number of Connected Calls is equal with the number of Received calls but less than the number of Attempted calls then the DUT cannot route all the calls The
199. lan on the Originate network traffic to satisfy the dispatching rules Scenario Execution Dial Plan sie Automatic Ts Cloud Codecs RTP Audio video Fax T 38 Fax T 30 SRTP Destination IFs None Source IPs The source IF addresses are taken From the associated Network see Traffic Network mapping tables in the test Phone numbers f Phone book entry Override phone numbers from destination activity f Phone book entry f User defined 1Fo 00000000 Use Tel URI parameters phone context example cam f User defined 1eofoo0n0000 Use Tel URI parameters phone context example com Phone book Edit verify all settings Restore defaults Figure 102 Emulated SIP Registrar dial plan 10 Define the test objective IxLoad uses a generic approach to determine the transaction rate with the Loops per second test objective When this test objective is used IxLoad instantiates all the channels or emulated endpoints and starts the call flow for the value specified in the test objective The test scenario is built such that one registration is performed per loop In this way the registration rate is directly controlled by the loop rate If the test scenario were used for a different call flow instant messages for example then the loop rate would control the instant messages rate PN 915 2611 01 Rev H June 2014 113 Test Case Determining the Maximum Transaction Ra
200. le However the required memory per translation is relatively small a few hundred bytes and today s devices are typically equipped with enough memory to support a large number of NAT translations NAT performance typically depends on several factors including The NAT mode NAT versus NAPT The type of application traffic The type of applications running concurrently The number of messages passing the NAT more messages results in more translations Translation Phases The translation phases of traditional NAT are as follows Address binding When an outgoing session originates from the private network the private address that initiated the session is bound to an external address the public address that is used to route the message further In the case of NAPT where many private addresses are mapped to a single public address the binding would be from the tuple of private address private TCP UDP port to the tuple of assigned address assigned TCP UDP port As with basic NAT this binding is determined when the first outgoing session is initiated by the private address private TU port tuple of on the private host Address lookup and translation After an address binding or address TCP UDP port tuple NAPT binding is established a soft state is maintained for each of the connections using the binding Packets belonging to the same session will be subject to session lookup for translation purposes Address unb
201. lick Port Assignments Add your assigned chassis 10 200 128 34 2 Identify your assigned cards and ports using the Annex provided 3 Add your first assigned port to the VoIP H 323Peer1 activity PN 915 2611 01 Rev H June 2014 66 Test Case Determining the Maximum Number of Concurrent Calls 4 Add your second assigned port to the VoIP H 323Peer2 activity Navigation 2 Pork Assignments sea Se eee Assigned Ports Test a a Chain _ Auto refresh status Chassis Chain Chassis Chain HG chassis 10 1 10 200 134 136 CM Card 1 10 100 1000 ELM 5T2 HHN Card 2 10 100 1000 CPM Ta ra Overview ge E Networks and Traffic E 8 New Traffic Flow El IT Originate NetTraffic Total Ports New Traffic Flow gk Traffict Met 1 of 1 ib Port 1 41 E DA Trafficz Net 1of1 Sa Traffici Metwork Bort Terminate DE Card 4 ACCELERON NP ME Port 1 5 1 A Trafficz Network E Port 1 4 2 TxLoad s2 554H Configured for 1 a E Port 1 4 3 xLoad S2 554H Configured for 1 E Port 1 4 4 xLoadfs2 554H Configured For 1 lt E Port 1 4 5 xLoad S2 554H Configured for 1 Port 1 4 6 xLoad S2 554H Configured For 1 E Port 1 4 7 xLoadfs2 554H Configured For 1 E Port 1 4 8 xLoad S2 554H Configured for 1 Port 1 4 9 xLoad S2y 554H Configured For 1 Port 1 4 10 IxLoad s2 554 Configured for 1 E Port 1 4 11 IxLoad s2 554 Configured for 1 o iE
202. lick the SIP_Callee NetTraffic and change the network settings accordingly to the particular test topology The number of IP addresses should match the number of channels defined in Test Objective 1000 in this example 8 Click the SIP_Proxy NetTraffic and for IP R3 the first IP range change the network settings accordingly to the particular test topology This will be the IP address of the emulated SIP server The same value should be set as the SIP Proxy IP address in the Caller and Callee activities The IP addresses of the other ranges under SIP_Proxy NetTraffic may remain unchanged while these are internal IP addresses used just to route the messages between the SIP Cloud to the SIP activities The number of IP addresses for the range IP R3 should be equal with the number of ports number of Ixia pots used to emulate the SIP Proxy 1 in this case The number of IP addresses of other ranges should be equal with the number of channels defined in Test Objective 1000 in this example Barro db 2 Trafficl Sd Caller nc SIP_Callee ope 3E Traffic3 vt Callee Ed SIP_Proxy F 32 Stack 2 gt Filter 3 TCP 3 Settings 3 gt GratARP 3 gt DNS 3 B MAC VLAN 3 A Ethernet 3 Unconfigured 40 40 50 1 16 0 0 0 1 Increment every IP R 4 Unconfigured IPw4 40 40 100 1 16 0 0 0 1 1000 0 0 0 0 0 0 0 0 Increment every IP R5S Unconfigured IPw4t 40 490 110 1 16 0 0 0 1 1000 0 0 0 0 0 0 0 0 Inc
203. licking Specify and entering the sequence 8718501 0001 4 Set Destination to Traffic2_Receive_Call 5060 by selecting the corresponding symbolic link from the list available for destination IPs PN 915 2611 01 Rev H June 2014 26 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Note When the number dialed must be different than the numbers specified in the remote activity the Override phone numbers from destination activity check box must be selected and a destination phone range must be specified Trafficl Make_Call VoIPSip Peer Source Destination IPs The source IP addresses are taken from the associated Network see Traffic Network mapping tables in the test Phone numbers Override phone numbers from destination activity C Phone book entry Use Tel URI parameters Phone book Verify all settings Restore defaults Figure 24 Dial plan settings Configuring SIP Parameters for the Make_Call Activity 1 Click the SIP tab 2 Set the SIP Port number to 5060 Note This parameter will also accept a range of ports for example 5060 6060 2 3 For back to back tests verify that the Use External Server check box is cleared If running against an external server use these settings instead at the bottom in the following figure e Select the Use External Server check box e Set the Server Address option with the IP address of the SIP Server e Set the Server Port option
204. lm test com User defined AVP s Subscribe to notifications of signaling path status Provision signaling flow information Initial provision of session information Figure 194 PCRF parameters PN 915 2611 01 Rev H June 2014 204 Test Case Measuring Quality of Experience for Voice Calls in LTE Note the PCRF implementations have variations in supported AVPs Attribute Value Pair To interoperate with a specific PCRF you may need to use custom AVPs in the Diameter messages on Rx interface This is possible in IxLoad by loading a file with the description of custom AVPs the field User defined AVPs allows this d Set the SIP Security parameters on the Server The SIP Cloud acts as the P CSCF a real P CSCF does the authentication of the SIP requests The emulated P CSCF can do the same functionality In the Security tab the type of requests that need to be authenticated can be defined as well as the Authentication algorithm and the credentials for each UE Select the Security tab enable MD5 Authentication Algorithm and disable AKA Authentication Click on Security Profiles Pool button Create a new profile and make sure you set the same values for the Phone Number Username and Password as for the emulated UEs on the originate side Edit Security Profiles tim Jetworks and Traffic TrafficFlow 1 User defined Wi VolPSipPeer1 iam Security subscribers count 64000 Fhone
205. ls aE Default Configuration Help Xs Preferences About Exit Figure 174 Option to Import a Compressed Repository File into IxLoad Create the Network Traffic for UEs 1 Open the IxLoad GUI 2 Add the Originate Net Traffic PN 915 2611 01 Rev H June 2014 191 Test Case Measuring Quality of Experience for Voice Calls in LTE The menu is context sensitive to access the Add Net Traffic button you have to select first the Networks and Traffic in the navigation pane B H gt tee IxLoaa oO itory n S1 USCA 200424 L Home Reports Traffic Start Apply Test1 a K2 Test Testi Config X Dut X aste Options Test Active Test impairment Networks and Traffic TraffiFlow1 on S OF os Stats Analyzer alll Test Overview Sacre Networks and Traffic izi g TrafficFlow1 ili Originate iii DUT iil Terminate hel Tia ia a Mele ansa Add Net Traffic Figure 175 3 Add the MME eNB 11 S1 U Stack Manager interface to the Originate Net Traffic By default the Net Traffic added in step 2 contains a range of IPv4 addresses The intention in this configuration is to emulate User Endpoints over MME and eNodeB that are connected through S11 and S1 U interfaces to the S Gateway These interfaces are provided by the MME eNB 11 S1 U stack manager component Select the Network right click on the IP 1 select Add above and then MME eNB S 11 S1 u Add Delete Add Stacks Stack Plugin
206. ls in LTE d Set the Audio parameters As on the originating side the terminating side must have the Audio settings on to play and receive audio Select the Audio tab enable the checkbox Enable audio on this activity set the Play for to 30 seconds and enable the Perform MOS checkbox xeaiton Dial Plan J SIP Automatic TS Cloud Codecs RIG Aud epic audio on this activity if unchecked all audio script functions will be SKIPPED Play Settings _ Enable jitter buffer Clip US_042 wav Format PCM Duration 32785 ms Size 524556 bytes Output level 20 dEm Play for dip duration or TalkTime all objectives except Channels a a a g Play for _ Perform Qov L Type Of Service TOS DSCP Q Perform MOS Calculate One Way Delay L Generate silence Null data encoded Comfort noise Verify all settings Restore defaults Figure 199 Audio Settings 16 Set the Timeline and Objective For the capacity test a test objective type will be set to Channels The specified number of channels will be concurrently active executing the call flow defined in the activities Set the Objective Value to 10 If you want to increase the test objective you will need to e Increase the number of Maximum Active UE Count in the User Equipment plugin under Network e Extend the sequence of Phone Numbers in the VolPSIPPeer1 dial plan e Extend the sequence of User names for the VolPSipPeer1 Authentication e Extend the s
207. lt destination domain name or host por Edit configurations Use Tel URI scheme For Source L Type OF Service Use Tel URI scheme for Destination Figure 148 SIP Server settings Note The server address and domain name has to match the ones configured on the gateway 6 Click the Audio tab to enable media and set the call duration Select the Enable audio on this activity check box click the Play for option set Play for to 50 seconds and select the Perform MOS check box With these settings the calls will have the duration of 50 seconds PN 915 2611 01 Rev H June 2014 158 Test Case Determining the Capacity of a VoIP to PSTN Gateway Trafficl SIPMakeCall VoIPSip Peer Scenario Exesuton ial Plan J IP Automatic TiS Cloud Codees RTP J Auto video Fax 38 Fox 1 30 SRTP Other Enable audio on this activity iF unchecked all audio script Functions will be SKIPPED Play Settings _ Enable jitter buffer Clip US_O042 way Buffer size 20 ms Format PCM Duration 32785 ms Size 524556 bytes bee compensation aa level 20 d m Max size L000 ms Max dropped consecutive packets Play For clip duration or TalkTime fall objectives except Channels beaks a C Perform Qo O ibe OF Service Units of Channels TOS IDSCP Class 1 0x20 deiei y Channel Selection First Channels Perform MOS Calculate One Wav Delay _ Generate silence Mull data encoded Comfort noise Figure 149 Audio setting
208. m were actually targeting governmental offices and agencies Most of these attacks are using open source software reprogrammed to perform floods of calls against a targeted pool of numbers while others have the capability to malformed the signaling traffic to compromise the recipient device PN 915 2611 01 Rev H June 2014 239 Test Case Telephony Denial of Service such attacks have been found and documented over time on security focused websites or third party security exploits tools however more and more of these have a security patch deployed Such malformed packet attacks typically generate system instability or exploit buffer overflows caused when receiving these frames Attacks could cause a remote DoS attack by triggering an undefined state of the SIP stack cause increase of time in the processing of new request allow unauthorized access to the resources or even allow remote execution of code Such attacks include INVITE of Death that uses crafted message to handle the content of Via header TDoS BYE attack that also affects the stability of the system by the use of crafted BYE messages and slow reply of SIP ACK with crafted SDP As an example an open source SIP proxy server has a serious vulnerability to large messages over TCP that may lead to a system crash The trigger to such behavior is a remote forged SIP REGISTER over TCP transport The stack allocates memory for the incoming packet and if the packet content information
209. mance o CPS versus signaling QoS charts CPS rate versus post dial delay post pickup delay call setup time and end call time CPS rate versus call establishment ratio and call completion ratio CPS rate versus retransmissions o CPS versus media QoS charts if applicable CPS versus MOS CPS versus PESQ LQ and PESQ LE CPS versus packet loss percentage and packet loss correlation PN 915 2611 01 Rev H June 2014 9 Test Case Determining the Maximum Call Setup Rate CPS CPS rate versus jitter and one way delay e To plot the performance characteristic of the DUT at least 10 data points should be included on the chart 5 percent 10 percent 25 percent 40 percent 50 percent 65 percent 75 percent 85 percent 95 percent 100 percent are suggested based on the maximum rate supported Using data points that are higher than 100 percent of the determined peak rate can be also useful because they can depict the behavior of the DUT when overloaded Some VolP devices only handle call signaling aspect while others handle both signaling and media simultaneously Further DUT performance characterization should be pursued by changing the test variables that influence the performance of the DUT Those test variables include e IP version IPv4 IPv6 e Transport layer UDP TCP TLS e P and port mapping for signaling and media 1 1 1 n e Call duration e Number of CODECs per user lf media traverses the DUT the negotiated CO
210. me e SSe SS565 fy 130 fen SSS 20 SSS SSS2eo ss a eae 200 ee ee 200 Call Setup Time E ee if ACK STOF an gt ACK l a Se gt ACK Vi gt lt STOP Both Way RTP Media CESS SS SSS SSeS SSeS SSeS SSeS SSS SS SSS SSS SS SSS SSS SSS S3 Figure 2 Call setup time for a call with authentication required e Post dial delay measures the time required to get the first ring back notification after the last digit of the destination phone number was dialed Tac UiS INVITE START gt gt Ay ll PDD ll ir 150 STOP gt lt Figure 3 Post dial delay PDD in SIP For SIP calls post dial delay measures the time from when the INVITE is sent until the 180 RINGING message is received The post dial delay has an important significance for the caller A large post dial delay is perceived as a Call that does not go through because the caller does not receive any PN 915 2611 01 Rev H June 2014 12 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems indication of call progress ring back tone A post dial delay value below four seconds is recommended for IP telephony systems When the post dial delay values exceed four seconds t
211. meters in the same test run You can create various scalability tests to measure the SUT s operating performance under actual telepresence traffic PN 915 2611 01 Rev H June 2014 232 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Table 84 Performance Variable Description Configure multiple Scale up the number of simulated channels to benchmark the maximum channels objective capacity that the system under test can handle Use single screen model By default the telepresence scenario uses a triple monitor camera environment Change the scenario to single screen emulated endpoint to simplify the test setup and decrease media traffic Change the video files In the Video configuration tab at the emulated endpoint side modify the playback file to various bitrates and codec modes Speaker rotation From Telepresence Settings configuration screen found in Video tab scheme modify the speaker duration time and rotation scheme This converges into different patterns of traffic through the SUT and offers valuable data for the QoS measurement Increase test time By increasing the play time for the audio and video streams the emulated duration call increases the overall duration This duration should be adjusted also into the Timeline and Objective s Sustain Time Troubleshooting and Diagnostics Table 85 Troubleshooting Solution Calls are not established Ensure that the SUT configuration is allowing SIP si
212. n ITU has ratified recommendations on how to measure it in the P 800 technical paper The methods of measurement are generally applicable regardless of the form of degradation the signal is suffering as it may be packet loss corrupted payload as an effect of bit errors various types of noises propagation delay and delay variations coding PN 915 2611 01 Rev H June 2014 222 Test Case Measuring Quality of Experience for Multimedia VoIP Calls and decoding schemes as well as the echo or side tone of an audio signal The score has the lowest quality rated at 1 with a maximum scale value of 5 To measure the user s rating for audio and video quality for telepresence applications IxLoad uses the industry s leading library from Telchemy For each individual channel if the MOS calculation option is enabled the agent performs a non intrusive assessment on quality displaying a wide range of statistics for channel monitoring and network diagnostics Additionally the Media Quality Assessment is performed using the VQmon engine for codec specific induced degradations due to variations in speech or video quality that each codec can have The VQmon library is running on all Ixia ports assigned for media traffic generation and it collects the required statistics about any network impairment such as packet loss buffer accumulation coding and transcoding times propagation delay and bandwidth capping to compute the user experience index for audio v
213. n conversation see prior section on the effects of delay The most common causes for jitter are network congestion and router switch queuing methods Packet Loss Packet loss is probably the most common factor affecting speech quality All voice CODECs can accept some packet loss without dramatically degrading speech quality Many CODECs supports as much as 5 percent of random packet loss while maintaining acceptable voice quality However loss of many consecutive packets dramatically affects voice quality even if the total percentage loss is low When designing networks and applications the target should be zero packet loss In converged networks where voice and data use the same resources voice traffic must be configured to have priority treatment over data traffic which is not as affected by delay jitter and packet loss High Availability Reliability Performance and Capacity Tests IP telephony networks satisfying the high availability requirement need to address consumers needs to register place calls and receive calls even at peak call rates In a similar way IP telephony systems must handle traffic even when devices are in maintenance or have failed High availability networks are assured by adding redundant systems for both signaling and media components Hence when a primary node in a voice network is down for maintenance or because of a failure a redundant device secondary or tertiary can take over the processing of
214. n the Net Traffic select the IP Mappings tab double click on the Distribution Group and select the IP Round Robin distribution type Networks and Traffic TrafficFlow1 v Terminate VoIPSipCloud 1 a VoIPSipCloud X Group name DistGroup1 Distribution Type IP Round Robin Description Distribution Type IP Round Robin Figure 183 Distribution Type of Network Ranges Add and Configure the SIP Peer Activities In this moment the configuration contains all the network elements The next steps will be to add the definition of the application traffic The intention of this configuration is to emulate SIP UEs on the eNodeB MME side that originate calls routed by the emulated P CSCF to emulated SIP endpoints placed behind the P CSCF This will be emulated by adding a SIP activity to originate calls on the MME eNB 11 S1 u network and a SIP activity to receive calls under the SIP Cloud PN 915 2611 01 Rev H June 2014 196 Test Case Measuring Quality of Experience for Voice Calls in LTE 10 Add a SIP peer activity on the Originate side Move the mouse over the Traffic element of the Originate network a plus sign will be shown click on it and select the VoIPSIP Peer activity Networks and Traffic TrafficFlow 1 t r a i Data Video E Voice Bulk MGCP Bulk SIP H248 MGC H248 MGW MGCP CA MGCP GW Traffici VoIPH323 Network Ranges By Port Dist aie Vo eNB
215. n the Scenario Editor toolbar For both scenario channel 0 and scenario channel 1 replace the Voice Session script object with the Multimedia Session script object the Multimedia Session script object allows both voice and video traffic to be simultaneously sent and received Leave the parameters of the Multimedia Session objects to their defaults the settings at the Activity level in Audio and Video tabs will be used Trafficl_Make_Call Scenario Execution Dial Plan SIP Automatic TLs Cloud Codecs RTP Audio video Fax T 38 Fax T 30 SRTP Other Dg x mm Ay check T E eeg H Bc dd Channel B Remove fpEcompact 4 Note Zoom 100 Workspace x Elow Scenario Channel 0 Activity Make_Call Traffic Traffict Hetwork1 Column Originate Link YolPLink1 B vor H323 Library H323 RAS Library Media Library ATP Generate DTMF ATP Detect DTMF ATP Generate MF ATP Detect MF ATP Generate Tone ATP Wait for Tone ATP Talk ATP Listen ATP Voice Session ATP Path Confirmation ALP D anken Ba 38 Fax session MGCP GW Library MGCP CA Library SIP Library SKINNY Library H 248 MGC Library H 248 MGW Library Figure 142 Test Scenario Editor with Multimedia Session PN 915 2611 01 Rev H June 2014 147 Test Case Using VoIP to Measure NAT PAT Performance 8 Select the Multimedia Session from Scenario Channel 0 and open its properties Right Click on object Object Pr
216. ne 2014 254 Test Case Telephony Denial of Service The following questions provide guidelines on how to recognize specific problems during or at the end of the test execution 2 Has the test objective been achieved Check the Call Rates view Statistic Name Loops Initiated per Second SIP Responses Received Table 89 Rate statistics Questions Have the loops been attempted continuously at a constant call rate during the Sustain Time How do the Loops initiated per second rate and the Responses Received rate compare to each other 7 Have any scenario loop failures been reported Check the Loops statistics view Table 90 Statistic Name Total Loops Successful Loops Statistics highlighting the pass fail result based on flow execution Questions 3 4 Failed Loops Aborted Loops Warning Loops Are the Successful Loops and Total Loops values equal Have any Failed Loops Aborted Loops or Warning Loops been reported Note failed aborted and warning loops highlight failures at the scenario level Consult the information found in the Event Viewer that will indicate the reason of failure and possible actions to remediate the fault Event Viewer VoIP SIP Error Events All Port Events 05 08 2014 05 38 38 0 05 08 2014 05 38 38 0 05 08 2014 05 38 38 0 05 08 2014 05 38 38 0 Total Events 1521899 Chassis Card Port 10 205 17 114 1 12 10 205 17 114 1 1
217. nels Play for Seconds _ Perform Qov L Type Of Service TOS DSCP Perform MOS Calculate One Way Delay L Generate silence Null data encoded Comfort noise Verify all settings Restore defaults Figure 191 Audio Settings PN 915 2611 01 Rev H June 2014 202 Test Case Measuring Quality of Experience for Voice Calls in LTE 14 Set the parameters of VolIPSipCloud1 activity Select the VolPSipCloud1 activity a Map the SIP cloud activity to the IP address defined for the Terminate Network In the Settings tab select the IP address from the list Only the Network Ranges with the Distribution Group Round Robin are shown Trafi oIPSipCloud 1 VoIPSipCloud Settings Pre jew Cloud Traffic Security Diameter IP Preference Only IPv4 only IPv6 ap Simulated SIP Servers Name IPAddress O o OS sip_server 1 i111 Figure 192 Map the SIP cloud to an IP address b Enable the Rx interface on the emulated P CSCF The P CSCF and the PCRF are connected through the Rx interface see Error Reference source not found Error Reference source not found The interface is used to confirm that call media requests conform to the appropriate policy open gates and pinholes in the media route specify the appropriate QoS request per flow charging information when needed and inform the P CSCF of media plane events The Rx interface uses the Diameter protocol and can be emulated by the SIP Cloud
218. ng the desired objective To achieve the desired call rate for a given number of simulated users the call hold time must not exceed certain limits IxLoad provides two working modes e Auto control of the media duration using the TALK TIME option that is available on all the RTP script functions for example RTP TALK RTP Listen RTP Voice Session e Script overrides of the TALK TIME duration This is a programmed duration constant across all calls which can be configured on the media script objects when set it overrides the TALK TIME value calculated by the test objective The Calls per Second objective can work in three different modes explained below The first mode allows the media duration to take assume the values calculated by the Calls Initiated per Second test objective In this mode the call rate is kept constant for the entire test duration Requirement examples e Attempt the calls using a steady call rate of 100 cps for 8 000 users e Attempt the calls using a steady call rate of 100 cps with a talk time of 3 minutes Calls Attempted per Second 5 Q D an D 2 5 D 2 D lt B T oO 30 Time hrs Figure 19 Example of constant call rate using Auto Control PN 915 2611 01 Rev H June 2014 22 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems The second mode is useful in some use cases when a very long conversation time is required for e
219. ngs can come from a number of different sources For example calls made over private data networks bypass PSTN network toll circuits avoiding regulatory and per call charges Many benefits are available using a common network for voice video and data applications The unification of voice text video and data on the same network allows quicker integration of multiple services improving the way users collaborate and leading to enhanced productivity Unified communications can transform a desktop application such as Microsoft Outlook into a communication center in which e mail voice and video calls screen sharing instant messaging voice mails presence and availability are easily accessible resulting in enhanced productivity As an example a meeting screen sharing audio can be recorded with a simple click of the mouse and then shared with team members who could not attend the meeting Voice mail notifications can pop up directly on the screen or be received via email User mobility and portability are advantages provided by IP networks Using software applications such as soft phones anyone can transform a laptop or PC into a mobile IP phone Hence employees can work from anywhere and home subscribers can gain additional mobility when they travel Regardless of whether they are traveling or working trom home the mobility aspect adds a simple yet powerful way to maintain communication User mobility allows IT administrators to pro
220. nnels span T1 ISDN PRI Out of band signaling 23 channels span T1 NCC No signaling E1 CAS In band signaling 30 channels span E1 ISDN PRI Out of band signaling 30 channels span E1 NCC No signaling 30 channels span There are several variants of ISDN and CAS depending on the type of interface The following protocols are supported in the current version of IxLoad Interface Type Type G704 Framing Format Line encoding HDB3 ISDN Signaling Protocol Protocol QSIG MFC R2 PN 915 2611 01 Rev H June 2014 160 Test Case Determining the Capacity of a VoIP to PSTN Gateway Parameter The variant type depends on the Signaling settings e For T1 E1 ISDN PRI NT network Variant termination TE terminal equipment e For T1 CAS FGB FGD Immediate e For E1 CAS None as If selected statistics for the range are computed and UDI MAUSNGS ENRON displayed in the StatViewer component of IxLoad PstnNetwork1 X12 Digital T1 E1 5 m cane name status e ol hace AA a i sman cenerate ms Detect ms Detect vs te PSTNDigitalRange 5 Unconfigured ama G704noCRC4 zm Figure 151 PSTN Network Settings Note All the parameters are consistent with the parameters set on the DUT it has the following settings Table 57 Variant NT If one device is set to Terminal the connected device has to be set to Network Note Multiple ranges can be configured for the same PSTN Network this all
221. nning the Test b g a Chassis Hide assigned ports EF Chain _ Auto refresh status Fa E T Chassis Chain EMG chassis 10 1 10 200 134 136 CMM Card 4 10 100 1000 ELM 5T2 CH Card 2 10 100 1000 CPM Te Ww Card 4 ACCELERON NP Uy ny neg eee eee generar om Card 5 ACCELERON NP senggeaee Port 1 4 2 IxLoad S2 554H Port 1 4 3 IxLoad S2 554H Port 1 4 4 IxLoad S2 554H Port 1 4 5 IxLoad S2 554H Port 1 4 6 IxLoad S2 554H Port 1 4 7 IxLoad S2 554H Port 1 4 8 IxLoad S2 554H Port 1 4 9 IxLoad S2 S554H Port 1 4 10 IxLoad s2 554 Port 14 11 IxLoad 5255A Port 1 4 12 IxLoad S2 554 Port 1 5 2 IxLoad S2 554H HE Port 1 5 3 TxLoad 52 554H LEE Fort 1 5 4 IxLoadiS2 S54H ef Port 1 5 5 IxLoad 52 S54H Configured For 1 Configured For 1 Configured For 1 Configured For 1 Configured For 1 Configured For 1 Configured For i Configured For 1 Configured For 1 Configured For 1 Configured For 1 Configured For i Configured For 1 Configured For 1 Configured For 1 Assigned Ports New Traffic Flow gk Traffict Met 1 of 1 Eha Trafficz Met 1of1 amp fi Port 1 5 1 Show IP Assignments Sssign PSTN Interface Figure 41 Port assignments page 1 Click Run to start the test execution 2 IxLoad will automatically display Statistic Views after execution starts Results Analysis The following ques
222. nterprise network SRC 192 168 1 100 5060 REGISTRAR amp PROXY SRC 172 16 100 9 5060 16 100 2 eo ee DST 193 16 148 244 5060 _ c SRC 75 83 202 1 6 5060 DST 172 16 100 2 5060 172 16 100 2 5060 1 REGISTER DST 193 16 148 244 5060 Alice 3 REGISTER 192 168 1 100 C D 2 REGISTER Session Border 168 1 Me Enterprise NAT NAPT Controller 172 16 100 9 Service Provider s WAN ii Ey Ni TARRE Core Network i 192 168 1 1 l 4 Nina 4 200 OK i GME 6 200 OK 75 83 202 16 5060 A 7 1025 5 200 OK l Media Gateway l 172 16 100 3 Carol 192 168 1 101 Figure 128 How a Session Border Controller resolves the NAT traversal issue for SIP messages When Alice s phone sends the REGISTER request it sends it to the public IP address of the SBC 193 16 148 244 port 5060 This address is provided by the service provider to configure the registrar server and outbound proxy server address on the IP Phone The REGISTER request is transmitted using the private address 192 168 1 100 from port 5060 The message 1 REGISTER has the following structure when issued by Alice s phone PN 915 2611 01 Rev H June 2014 138 Test Case Using VoIP to Measure NAT PAT Performance 192 168 1100 5060 193 16 148 244 S060 REGISTER sip myserviceprovider com SIP 2 0 Via SIP 2 0 UDP 192 168 1 100 5060 From Alic
223. oice quality Setup The EPC isolation test configuration will be used for this test where IxLoad will emulate e the User Endpoints over eNodeBs and MME the left side of the topology diagram in the the below figure e the IMS network that is the P CSCF and all the devices behind it Voice calls will be originated from the UEs eNodeB MME and terminated by user agents behind in the IMS network PN 915 2611 01 Rev H June 2014 190 Test Case Measuring Quality of Experience for Voice Calls in LTE s PORF DUT 7 i l t Gx M Rx P CSCF MGW IMS Network i gt i q SGW DUT P GW DUT Figure 173 EPC Isolation Test Configuration Step by Step Instructions The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test The final IxLoad configuration as a result of these steps is provided on the blackbook ixiacom com Web site see VolP VoLTE IxLoad6 0 crf The configuration consists in two files the test repository a file with rxf extension and the test scenario for the SIP call flow a file with the tst extension These two files are archived in a single crf file To import a Compressed Repository File crf in IxLoad use the command Import under the File menu d gt a Save As l Save Ss Open Repository New 2 Export amp Import Too
224. ok entry W Phone book entry User defined 2001 2010 gt User defined _ Use Tel URI parameters Verify all settings Restore defaults Phone book Figure 196 Dial Plan b Enable this activity with the SIP Cloud The VolPSipPeer2 activity emulates SIP User Agents in the IMS core behind the P CSCF that is emulated by the SIP Cloud activity That means the all the traffic between EPC and the IMS core goes through the P CSCF VoIPSipCloud1 activity that will receive the SIP Requests and will forward them to the User Agents behind it PN 915 2611 01 Rev H June 2014 206 Test Case Measuring Quality of Experience for Voice Calls in LTE Select the Cloud tab enable the SIP Cloud simulation checkbox enable Virtual IPs checkbox and a range of 10 Virtual IPs with the staring IP Address 23 23 23 1 Traffic VoIPSipPeer VoIPSip Peer atc Tg Cloud Oadecs RTP Audi Video Fax 38 Fax 30 SRT yimulation using settings from VolPSipCloud1 Cloud topology Outside World ese Used Simulated SIP Servers sip server 1 T Simulated SIP Users Enable Virtual IM Virtual IP IPy4 Dispatching rules Figure 197 Associate a SIP Peer Activity with a SIP Cloud activity to emulate SIP User Agents behind a SIP Proxy There is no need to set the SIP Server IP address or the Credentials for Authentication for this activity as it communicates internally to the emulated P CSCF
225. one it acts as a SIP user agent or endpoint for the registrar server located in the Service provider s network Hence the SBC separates the communication in two distinct SIP dialogs one with the phone and one with the registrar server We will refer to this role as a back 2 back user agent B2BUA To address the NAT traversal issue the SBC detects the presence of a basic NAT by comparing the IP address and port number set in the Via headers with the source IP and port number set in the IP header Instead of responding with 200 OK to the address set in the Via header it sends the response to 75 83 202 16 port 1025 which represents the public address of the CE NAT gateway that issued the REGISTER request The SBC also adds the Expire header in the response which controls how often the endpoint will re register The original IP and port will be re used inside the payload PN 915 2611 01 Rev H June 2014 139 Test Case Using VoIP to Measure NAT PAT Performance Because the SBC needs to register as an endpoint with the registrar server using the private interface that connects the SBC to the registrar server it also uses a NAT translation that maps the public address 192 16 148 244 5060 to its private address 172 16 100 9 5060 The NAT translation also includes the Expire value which will be updated upon receipt of 200 OK response from the server The following figure illustrates the path of the 200 OK response from registrar serv
226. onfiguring and determine the QoS for telepresence using IxLoad Video conferencing has played an important role in bringing additional value to communications by allowing more interaction between multiple participants Key benefits include an increase in productivity frequent and fruitful interaction between staff in different locations a better information flow and improved decision making information sharing and knowledge from specialists in different branches all in one delivering a balanced quality of life for the business environment PN 915 2611 01 Rev H June 2014 213 Test Case Measuring Quality of Experience for Multimedia VoIP Calls There are several challenges that this technology brings from technical aspects such as propagation latency device processing capabilities and solution convergence in terms of standardization compliance Starting from the initial point where the media is captured several enhancements are done to improve the QoE as follows Advanced signal Detection Voice activated switch that helps determine the media delivery For example the speaker or the room with the loudest sound will be the one seen by all the other participants Echo cancelation A reflected source of interference created by the original signal source The device detects the utterances that re enter the audio path that comes out of the videoconferencing codec output with some time delay If untreated it can result in hearing your own vo
227. operties to configure the video clip and audio clip used for multimedia transmission a Select the Play Audio Clip and the Play Video Video to be transmitted all video properties will be retrieved from the selected video clip for example CODEC bit rate profile level Multimedia Session Properties Video Play Parameters advanced Playback Settings Output Settings Delay Before Execution Static Expression v 0 ms IV Play Audio Clip US_042 way ace Output level dB pon ne Dee ie S2956 bytes Time 32785 ms s RTP Audio video Fax 7 38 Fax T 30 SRTP IV Play Video Video MufasaSimba mp4 is Cy Check a g Format Custom 412 x 240 Profile Main ED Levet 1 3 Duration 126900 ms 4 Note Zoom 1 Resource Pool Management Ctrl L Terminate Conditions I Stop playback on first detected Plays one or more WAVE Files and or MP4 files The wave files can have the a Following characteristics Coding PCM A LAW MU LAW Sampling frequency 8 KHz Additionally the WAVE files must be Mono and v Restore Defaults OK Cancel Figure 143 Multimedia Session left and Resource Pool Management button right b New audio and video sample files can be added by clicking Resource Pool Management available in the Scenario Editor toolbar Configuring the Settings for Video Traffic 1 Click the Make_Call activity Click the Video configuration tab 2 3 Enable the video traffic by selecting t
228. other RTP MOS Worst How does the RTP MOS Worst score compare with the max theoretical score for the CODEC used PN 915 2611 01 Rev H June 2014 185 Test Case Determining the Performance of a Session Border Controller RTP MOS Instant Are any times without an instantaneous MOS value Best Avg Worst How frequent are the changes in the instantaneous MOS values RTP MOS Per Call Howdo the MOS per Call statistics compare with the RTP Best Avg Worst MOS Best and RTP MOS Worst statistics Table 70 Basic RTP QoS statistics see RTP QoS and RTP Advanced QoS statistics views Statistic Name Questions RTP Packets Sent 1 Are there any differences between RTP Packets Sent and RTP Packets Received RTP Packets Received RTP Packets Lost 2 Does the difference match the value of RTP Lost Packets RTP One Way Delay us 3 Is the One Way Delay higher than 100 ms RTP Delay Variation Jitter us RTP Interarrival Jitter us 4 What is the max Delay Variation Jitter 5 What is the max Interarrival Jitter Table 71 RTP Jitter distribution statistics Statistic Name Questions Packets with Delay Variation Jitter up to 1 m Packets with Delay Variation Jitter up to 3 Packets with Delay Variation Jitter up to 5 m m m Packets with Delay Variation Jitter up to 20 m Packets with Delay Variation Jitter up to 40 m Packets with Delay Variation Jitter over 40 ms Table 72 S S S l l l Assuming
229. oud simulation using setting from check box is selected 5 Check whether the dispatching rule in Override default dispatching rules has the proper sequence of phone numbers in the Formula for dispatching field 5552 0000 in this example Setting the Wait_Call Activity Parameters The calls between SIP User Agents Caller and Callee via the SIP Proxy are composed by two call legs one between Caller to Proxy and one from Proxy to Callee The SIP proxy server is emulated by two activities behind the Cloud Wait_Call and Make_ Call Wait_Call terminates the first call leg from Caller and Make_Call originates the second call leg to Callee The two activities on the SIP Proxy Wait_Call and Make_Call communicate between them to synchronize the call flow and to pass parameters from one call leg to other for example destination phone number and SDP information 1 Click the Wait_Call activity under SIP_Proxy NetTraffic 2 Check the Scenario it contains a set of procedures to receive the call send the proper responses and handle the end call sequence 3 Click the Dial Plan tab and set the Source Phone Numbers 5552 0000 in this example this sequence should match the one specified as Destination Phone Numbers for the SIP_Caller activity this is the criteria for matching the incoming SIP Register messages to this activity 4 Click the Cloud tab and ensure that the Enable SIP Cloud simulation using setting from check box is sele
230. ource Modify the header as Contact lt sip VOIP_Phone 127 0 0 1 gt The variables usage is syntax sensitive so attention should be given when composing the field An invalid syntax should trigger a SIP parser error on the DUT and this represents another test case Re Send RE ag Ww eh Error Figure 227 Accessing the lower level Registration procedures PN 915 2611 01 Rev H June 2014 242 Test Case Telephony Denial of Service REGISTER AUTO REQUEST URI SIP 2 0 Via AUTO VIA From AUTO FROM To AUTO TO Call ID AUTO CALL ID AUTO CSEOQ E lt 41 ax Forwards Content Length AUTO CONTENI LENGTH Expires 3600 Edit Options Change case Create From Template Load From File Modified by user Iv Case sensitive Delay before execution Static Expression fo mg Message body Send audio SOP Offer Send custom message body Edit Custom Send a sip request message Figure 228 Modifying the Contact header information PN 915 2611 01 Rev H June 2014 243 Test Case Telephony Denial of Service 7 Inthe Behavior tab of the Send Request Properties enable the checkbox for Extended Variables Support This enables the endpoint to evaluate the VOIP_Phone as a variable and construct the message using the phone numbers configured in the Dial Plan configuration tab This checkbox should be enabled whenever variables are used and their values should be used in the SIP message P
231. outgoing packet the HTTP GET As the CE router receives each packet on its private interface it removes the source IP address 192 168 1 100 from the IP header and replaces it with its own public IP address 75 83 202 16 while maintaining the source port number and then calculates and modifies the IP and TCP checksum No changes are made to the payload After those operations complete the packet is sent to the PE router 75 83 200 1 which further routes the packet towards the initial destination 209 132 176 30 22 80 When the HTTP server located in the external domain redhat com receives the GET request it replies with a 200 OK response which is sent back using the source 209 132 176 30 22 80 and with the IP destination Modification of the IP Header for the INCOMING packet response The IP Header of the response packet as it is received on the public interface _ pone w naa mom a Translation 75 83 202 16 192 168 1 100 4 Private Public Interface Interface ri The IP header of the modified HTTP response as it is generated on the private interface Figure 109 Modification of the INCOMING packet HTTP Response In this example we see how NAT works by substituting the source IP address from the private network one private address in our example 192 168 1 100 with a public IP address in our example 75 83 200 1 lf the second host 192 168 1 200 from the private netwo
232. ows creating multiple PSTN activities on the same Net Traffic each activity with its own network parameters Add a PSTN Activity 1 Inthe PstnNetwork1 Traffic2 click the button to add an activity PN 915 2611 01 Rev H June 2014 161 Test Case Determining the Capacity of a VoIP to PSTN Gateway 2 Choose the Voice gt PSTNDigital Peer activity type the activity PSTNDigitalPeer1 will be added a Terminate 3 ap a Traffic2 Most Recently Used E E Terminate Yoj Figure 152 Edit the Call flow for PSTNDigitalPeer1 Similar to the advanced VolP protocols for example SIP Peer the call flow for PSTN activity is defined by one channel of a test scenario The same test scenario may contain VolP and PSTN channels in this way the actions on one activity may be synchronized with the actions on the other activity This example uses a single test scenario for SIP Make Call activity and for PSTN Receive call activity The scenario for SIP Make Call was created at the Step 3 of Add and Configure the SIP Peer Activity The steps to add the second test scenario channel to the existing one and associate it with the PSTN activity are as follows 1 Click the SIPMakeCall activity 2 Click the Scenario tab PN 915 2611 01 Rev H June 2014 162 Test Case Determining the Capacity of a VoIP to PSTN Gateway 3 Click Add Channel A new channel is added to the test scenario and a window to map the new cr
233. packet 4 209 out of 3 079 504 This is 0 1 percent of the total number of packets a small amount that cannot cause a significant degradation of quality of voice Packet loss may not affect all the calls some calls may be unaffected while others can have high level of packet loss PN 915 2611 01 Rev H June 2014 99 Test Case Subjective Quality of Voice In this case the Maximum MOS and Maximum PESQ values are equal to the ideal values and the Average MOS and Average PESQ values are close to the maximum values This is an indication that only some calls are affected by packet loss This is confirmed by RTP statistics per channel PE aan esi ace ea ar i ia Fil nT ia seep 7 RTF fvolPSip Packets Sent 2 766 612 2 885 426 2 974 697 3 034 819 3 073 629 3 079 504 3 RTF fvolPSip Packets Rec 2 762 291 2 881 077 2 970 315 3 030 364 3 069 027 3 074 946 3 RTP VoIPSip Bytes Sent 171 730 928 126 958 744 130 886 668 133 532 036 135 239 676 135 498 176 135 RTP volPSip Bytes Recei 121 540 804 126 767 388 130 693 860 133 336 016 135 037 188 135 297 624 135 RTF fvolPSip Throughput 35 305 25 969 24 880 18 670 7 211 0 RTF fvolPSip Throughput 35 251 25 948 24 825 18 667 7 183 42 RTP f VoIPSip Tx Packets T T T T T T 4 209 4 209 4 209 4 209 4 410 4 546 PIF LvolPsip Maximum ao RTP VoIPSip Packet Error 0 0 0 0 0 0 Figure 84 RTP QoS in case of packet loss T
234. port emea ixiacom com 44 1628 408750 online support form http www ixiacom com support inquiry location em ea Support Support Field Asia Pacific ixiacom com 1 818 871 1800 Option 1 online support form http www ixiacom com support inquiry 25
235. r voice traffic RTP RICP audio The test methodology Test Case Determining the Maximum Call Setup Rate CPS uses a SIP configuration example with a CPS objective In this test we will reuse the same configuration and we will modify the test objective to allow us to sustain a number of concurrent calls where voice and video traffic are both enabled The configuration changes required to enable video traffic are common for SIP and H323 traffic PN 915 2611 01 Rev H June 2014 145 Test Case Using VoIP to Measure NAT PAT Performance Setup In this test topology an Ixia port emulates a private network consisting of VoIP clients that establish calls through a NAT enabled device e g ALG or SBC with the signaling server and media gateway located in the public network Public IPs and ports Figure 140 Test Topology Step by Step Instructions The final IxLoad configuration as a result of these steps is provided on blackbook ixiacom com Web site see IxLoad 5 10 Voice SIP NAT crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test Depending on the protocol used SIP or H 323 start with the configuration steps highlighted in Determining the maximum call setup rates SIP example and Determining the ma
236. rder Controllers From the NAT point of view the SBC performs the role of an ALG translating the addresses and ports in the application data between private and public addressing schemes An SBC can be divided in two logical components e A Session Border Element that handles the signaling part of a VoIP call and PN 915 2611 01 Rev H June 2014 137 Test Case Using VoIP to Measure NAT PAT Performance e A Data Border Element that handles the media traffic associated with a call Those components can be collocated or it can be distributed From the deployment point of view the SBCs can be seen in five common scenarios e At the border between two service providers e At the border between a service provider and their customers e As a central media transcoder e Between different VPNs provided by a service provider e As a network resource controller By rewriting the IP addresses and ports included in the signaling headers the SBC allows VoIP traffic to be transmitted and received from a device behind a firewall NAT without requiring the customer s firewall NAT to perform that function Registration To place and receive calls Alice s phone must register with the registrar server located in the service provider s network The service provider placed a session border controller at the edge of his network to address the NAT traversal issue His enterprise customer is not required to upgrade his NAT Firewall device at the edge of the e
237. re 40 simulated user agents that originate SIP calls There are 760 Attempted Calls each simulated user agent attempted 19 calls during the test duration The number of attempts depends on the call duration the Talk Time in the Voice Session function The number of Connected Calls is equal to the number of Attempted Calls indicating that there are no problems with the call setup and the DUT can handle this volume of calls in the presence of video and data traffic The number of End Calls Completed is 1 520 double the number of attempted connected calls because both call legs are successfully ended Other statistics that should be checked in addition to the number of calls completed and the RTP MOS e Number of retransmissions open the SIP Messages view to see the Retransmitted Msg Statistics no retransmission should occur e VolP SIP Errors no errors should occur e RTP Streams the number of calls with incoming RTP should equal the number of completed calls The number of calls without incoming RTP should be zero e Lost Packets should be close to zero less than 0 1 percent of the total RTP packets e One Way Delay Max should be less than 100 usec This should be checked against DUT specifications PN 915 2611 01 Rev H June 2014 85 Test Case VoIP Quality of Service in Converged Networks Troubleshooting and Diagnostics Diagnosis Suggestions The target throughput is not If the device is not reachin
238. re managed by individual enterprises for own communication purposes or is deployed as a service solution for high end communication Multipoint Teleconferencing requires a dedicated network design to preserve the QoS that the platform promises to deliver A basic business to business call follows the following flow The flow differs from one vendor to the other since they adjust the call flow depending on the device technical capabilities ie Telepresence Service Telepresence Endpoint Server Endpoint Device Registration Device Registration a l Initial Call Request l In Call Media Negotiation Real time Media Flows Call Termination Figure 204 Business to business telepresence simplified call flow The signaling used by the telepresence application is Session Initiation Protocol SIP This is a widely deployed signaling protocol for most communication sessions such as voice and video over Internet Protocol Endpoints should perform registration to comply with enterprise s security policies This is done in various ways to allow information exchange only with the trusted entities The contact header from the SIP REGISTER provides the necessary details for connectivity to the endpoint such as IP address transport protocol port number and endpoint extension After successful registration the endpoint is ready to initiate or receive calls The user can select a predefined entry and trigger a telepresence in the user
239. reen in high definition using the 1080p resolution This represents intense bandwidth consumption and assists in qualifying the system under test for media delivery The video stream from the template is H 264 encoded with an average bandwidth of almost 2Mbits The video file payload can be changed for the video configuration tab from the Playback Clip option There are several files encoded on different bandwidths and resolution as part of the IxLoad installation and the user has to option to load custom files in the media library by accessing the Media Setting icon on the configuration ribbon Home Reports Views Traffic A o s A i A 4 1B Apply telease Network Traffic ScriptGen Media Config _onfig Wizards Wizards Settings Tests Wizards Tools Figure 217 Accessing Media Settings options PN 915 2611 01 Rev H June 2014 228 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Playback erform MOS Clip TIP_HD7_1920 1072 CAVLC _ Accept stream SSRC changes Ignore Hint Track Codec H264 Duration 10000 ms Size 2431343 bytes Bitrate 1897 kbps _ Single NAL Unit _ Play for dip duration or TalkTime all objectives except Channels STAP A with FU A fragmentation Play for ECONS SVC Settings _ Conference mode 6 Indude PACSI NAL Telepresence ee x Telepresence Settings E Use Single NAL unit _ Type Of Service TOS DOSCP Verify all settings Restore defaults Fig
240. rement every IP R 6 Unconfiqured IPw4 40 40 120 1 16 0 0 0 1 1000 0 0 0 0 0 0 0 0 Increment every 5 el TF 7 InraonFin rer TP 40 40 130 1 1A 0 A A d Tn AAA A Tocrement every Figure 167 Set the IP address of the emulated SIP Server PN 915 2611 01 Rev H June 2014 177 Test Case Determining the Performance of a Session Border Controller 9 Click Traffic2 under SIP_Proxy NetTraffic and check whether each activity has the SIP traffic mapped to one and only one network range Check that the Distribution Group for the range used by SIP_Proxy activity the cloud is Round Robin 4 7 SIP Caller oie 2 SIP Prox BB ee sr Prony car 3E Traffict ey Caller s a Tiamo 2 Traffic3 ey Callee Ed Trafficz User Source IP Mapping Activity User Source IP Rule per pork R_ Caller Use Consecutive IPs IP Mappings f Command Editor petore anaes my Port pistroutioncrow ppsa ee el TT eee E IP 3 Groupi Consecutive IPs 7 m L aal m m m m E Network Range IP R4 in SIP_Proxy 40 40 100 1 1000 ia i l Ll Network Range IP R5 in SIP_Proxy 40 40 110 1 1000 i ia L L Network Range IP R6 in SIP_Proxy 40 40 120 1 1000 Ll Ll Ll L Ll Network Range IP R in SIP_Proxy 40 40 130 1 1000 a a L L L L m m m m L m m Network Range IP R3 in SIP_Proxy 40 40 50 1 1 L Ll Ll L Figure 168 SIP activities mapping to IP ranges for emulated SIP Proxy Setting the SIP_Call
241. rent configuration The activity playback is set for Telepresence mode To measure the quality of video for the emulated traffic enable Perform MOS and Calculate One Way Delay options The template configuration emulates a Teleconference session with the duration of 60 seconds There are three emulated speakers each talking for 20 seconds The telepresence speaker rotation scheme can be changed from Telepresence Settings as well as other protocol specific options PN 915 2611 01 Rev H June 2014 22 Test Case Measuring Quality of Experience for Multimedia VoIP Calls tix Telepresence Settings Endpoint Type Resolution L_ Support Legacy Video _ Single Screen C 720p 1080p _ Support Legacy Audio _ Support Audio G 722 Legacy _ Audio Activity Metric Conference duration 50 seconds _ Audio Dynamic Output Channels Speaker rotation scheme Sequential _ Video Refresh Flag ee ES rs _ H 264 Inband Parameter Sets SPS PPS _ H 264 arithmeting coding CABAC Lise presentation stream _ H 264 Long Term Reference Pictures LTRP Audio dip UK 32 Telepresence48 mp4 _ H 264 Gradual Decoder Refresh GDR Video clip TIP_Presentation_ 1024 768 _ H 264 High Profile 8x8 transforms ere a Seconds _ Unrestricted media Duration 10 Seconds Use RTCP Feedback for Video Presentation rotation scheme Sequential Figure 216 Telepresence settings options The simulated endpoints act as a triple sc
242. rk initiates a similar request while the transaction from first host is still alive a secondary public IP address will be required Because with basic NAT only the IP addresses are substituted a public IP address must be available for each concurrent session PN 915 2611 01 Rev H June 2014 122 Test Case Using VoIP to Measure NAT PAT Performance Because all the packets initiated by the CE router will replace the original source address with its own public address the external network will see the entire private network as a single device the CE router with the IP address 75 83 202 16 22 as shown in the following figure External Network public ZR NAT NAPT 75 83 200 1 22 75 83 202 16 22 Figure 110 How the public network sees the NATed network This example also demonstrates how NAT provides a privacy mechanism by hiding the private network from everyone in the public network Routers and firewalls use NAT as a layer of security because a communication pinhole between the private IP and the public IP of the router firewall is created only when a solicited request is received that is a request or connection initiated from the private network An unsolicited request that is a request or a connection that is initiated from the public network is by default dropped because no communication path exists This behavior has the disadvantage of taking away the end to end significance of an IP address which can disabl
243. rminator of the call In addition to the call setup messages the DUT has to transmit the Registration requests from the SIP User Agents to the SIP Registrar server The typical SIP call flow for Registration and Basic Call is as follows Alice s al softphone phone REGISTER 200 OK REGISTER 00 OK IN ITE 100 Trying IN ITE 100 Trying 180 Ringing 180 Ringing 200 OK 200 OK ALK ALK d Media Session E ETE ETE 200 OK 200 OK Figure 162 SIP Call flow for Basic Call via a proxy over an SBC PN 915 2611 01 Rev H June 2014 173 Test Case Determining the Performance of a Session Border Controller Objective This test determines the maximum rate of SIP calls that can be established through a DUT between endpoints in the Private network when the SIP Proxy is in a Public network Setup In this test topology a pair of Ixia ports emulates SIP User Agent in a private network and the third Ixia port emulates the SIP Registrar and Proxy server in the public network The SIP User Agents register to the emulated SIP Register server and establish calls through the emulated SIP Proxy server over the DUT an Application Layer Gateway or a Session Border Controller Private Network IXIA s Emulated SIP UAs gt IXIA IXIA Emulated SIP Registar and Prox Private IPs and ports Servers ey ener erent Figure 163 Test Topology for SIP user agents and SIP Server emulation Step by Step Instructions
244. rvice Server All endpoints joining the video conference will call the Service Server it will relay all signaling and media streams from to all participants in the conference From a simplifies perspective the multipoint telepresence conferences are not more than several point to point calls all landing to the same endpoint the difference being the dialed number destination is a Service Server The signaling and media negotiation are otherwise the same and follow similar rules Depending on the implementation of the Service Server the SDP negotiation with each endpoint is done at the maximum capabilities of the endpoint or at the level of the least capable endpoint In the former case the Service Server will do the transcoding in the latter case all devices will be exchange media at the same level of quality as an example if a multipoint telepresence call has negotiated all the endpoints at 1080p resolution with Best quality for video and a new endpoint joins that maximum capabilities are at 720p resolution with Good quality the SDP information will be negotiated independently to match the announced capabilities PN 915 2611 01 Rev H June 2014 216 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Telepresence Service Telepresence Endpoint Server Endpoint Device Registration Device Registration Initial Call Request Initial Call Request rr In Call Media Negotiation In Call Media Negotiation Real time Media Flows Real
245. rying 3 00 00 30 960368 NE SIP 2 0 401 Unauthorized 4 00 00 30 961693 L REGISTER sip 20 1 100 1 SIP 2 0 5 00 00 30 962481 SP 0 100 Trying ee 6 00 00 30 962577 t A A rn 200 OK Figure 99 SIP Registration traffic captured on an emulated SIP Endpoint If the use case requires emulation of a SIP registrar server without Authorization then the test scenario should be changed On Channel 1 which is the call flow of the SIP Registrar server connect the Ok output of the first Send 100 Trying function to the Send 200 Ok function In this way the Send 401 Response and Wait for a second Registration request functions are skipped 6 Define the Dial Plan and destination IP for the emulated endpoints the Originate network traffic The source dial plan should be configured according to the test procedure specifications The destination IP address is not used in this type of test Scenario Execution Dial Plan siP Automatic TLS Cloud Codecs RTP Audio video Fax T 38 Fax T 30 SRTP Source Destination IPs The source IP addresses are taken from the associated IPs None v Network see Traffic Network mapping tables in the test Phone numbers Override phone numbers From destination activity C Phone book entry C Phone book entry User defined 160 00000000 User defined 1 70 00000000 Use Tel URI parameters phone context example cor
246. s PN 915 2611 01 Rev H June 2014 72 Test Case Determining the Maximum Number of Concurrent Calls Table 23 Distribution of RTP Consecutive Lost Packets Statistic Name VEUT E Questions maae oo Packet Sequence Three Packet Sequences Consecutive Loss of Four or 1 Assuming that packet loss was reported what ae 1 is the distribution of the lost RTP packets faces Packet Sequences Consecutive Loss of Eleven or oO More Packet Sequence Table 24 RTP Streams Statistic Name VEUT E Questions What is the max number of concurrent RTP Streams reported by IxLoad For what duration does the Concurrent RTP Streams indicate that all the channels were active Number of calls with incoming SS _ Are any calig without RTP RTP packets Are any calls with RTP Is this number equal to Number of calls without incoming RTP packets 2 x Calls Connected Troubleshooting and Diagnostics Concurrent RTP Streams Concurrent RTP Streams max The following table summarizes some of the common issues that may be encountered running a Capacity test Table 25 Troubleshooting Solution The Incoming RTP throughput is not Check the reported RTP Packets Lost RTP Consecutive constant during the Sustain Time or Packets Lost and compare the RTP Packets Sent with RTP has lower values than the Outgoing Packets Received This issue may be caused by a large RTP Throughput number of packets being dropped or by calls without RTP pac
247. s P a in Pl Build Stack S CO a Stats Analyzer alla Test Overview wes S34 Networks and Traffic amp treffictiow1 ili Originate iii Dut Terminate Ed Timeline and Objective Ports QuickTests ca X a G X aia Delete r TEAL Load VoLTE UE amp P CSCF nf S2 USCA 200424 Home Reports Views Traffic Network a Q E FEH Grid Selection Mode Plug in Grid Operations 7 Settings X Delete Range s ait Networks and Traffic TrafficFlow1 ugin Grid F z a 7 Networki A E Traffici zamm Networki gt Settings 1 E GratARP 1 E DNS 1 p Filter 1 p TCP 1 2 eal Ar e r Ethernet 1 ed Add above 2 b ae z 3 MME eNB S11 S1 U E4 Gateway Gateway Incremer Rename 100 0 0 0 0 Import Overwrite EAROUDY POPS SSL VPN Server DSLite PGW S5 S8 6rd Llog Event Viewer See ae l 7 a Test2 Unconfigured 00 00 00 Webu 00 00 00 Overload Protection On eNodeR Lu y Figure 176 PN 915 2611 01 Rev H Add MME eNB 11 S1 U Stack June 2014 192 Test Case Measuring Quality of Experience for Voice Calls in LTE 4 Configure the IP addresses for the emulated eNodeB and emulated MME a One eNodeB with the IP address 30 0 0 1 is emulated by this configuration Select the IP ENB element and set the Address and the Count User Equipment FEA
248. s S CSCF controls over the Mr or Mg interfaces the Media Server and Media Gateway for voice routing PN 915 2611 01 Rev H June 2014 189 Test Case Measuring Quality of Experience for Voice Calls in LTE PCRF DUT MME _ we ee ION i s i N j Signaling EX ar a 4 i mm 5 Legacy ea Networks GSM PSTN Mobile Phone Media Server Media Gateway Figure 172 VoLTE Topology To assure a good quality of voice a dedicated bearer with high QoS Is used for conversational speech in the EPC domain The allocation of the dedicated bearer is requested by the P CSCF to the Policy and Charging Rules Function PCRF over the Rx interface this is a Diameter interface SMS over IP is also a functionality specified by VoLTE The UE submits a short message via a SIP MESSAGE request that follows the same path to the S CSCF from here depending on the user profile obtained by S CSCF from HSS the SIP request is sent to the IP SM GW IP Short Message Gateway for simplicity this server is not represented in the topology shown in VoLTE Topology figure The submission report is sent by the IP SM GW to the UE as a SIP MESSAGE Request The SMS submit and submission report requests use the same SIP Method but with different content body Objective The goal of this test methodology is to determine the capacity of the EPC to handle a specific volume of calls without degradation of v
249. s the Overwrite phone numbers for destination activity check box has to be selected SOURCE Destination IPs The source IP addresses are taken From the associated IPs None Network see Traffic Network mapping tables in the test Phone numbers Override phone numbers From destination activity O Phone book entry C Phone book entry User defined 160 00000000 User defined 717444 0001 0030 Use Tel URI parameters _ Use Tel URI parameters Figure 147 Dial plan for the SIP activity Note Same dial plan has to be configured on the gateway 5 Click the SIP tab to configure the destination of SIP messages Select the Use external server check box and set the Server address to 192 168 1 100 and Domain name or local IP to voice ixiacom com Traffic SIPMakecCall VolPSip Peer Enable signaling on this activity SIP Part S060 iF unchecked all SIP script Functions will be SKIPPED Transport settings Use external server Maximum message size on UDP 1024 Server address 197 168 1 100 _ Override transport specified in scenario Server port 5060 Domain name or local IF VOICE Miacom com TCP send immediate C Qutbound prozy Enable FQDN resolution _ Registrar server Authentication UAC User name Anonymous Construction of SIP messages Password AKA authentication settings C Override default contact settings Select configuration lt None gt C Override defau
250. s for SIP activity Add and Configure a PSTN Net Traffic 1 Click Add Traffic Flow Element under Terminate and then click PSTN Net Traffic EH Add Traffic Flow Element wy ee oar b a Traffic3 gt Terminate NetTraffi a8 Terminate Sig ivi PSTNDigitalPeert etTraffic ee Subscriber DLT PSTH NetTratFic Figure 150 Adda PSTN Net Traffic PN 915 2611 01 Rev H June 2014 159 Test Case Determining the Capacity of a VoIP to PSTN Gateway 2 Click PstnNetwork1 and edit the network parameters Table 56 Parameter WEES Notes The type is selectable between E1 or T1 Important All board spans of the same board must be configured in the same way either T1 or E1 mixed configurations of one board spans are not allowed The T1 E1 framing format can be either of the following e For T1 D4 Superframe default or ESF Extended Super Frame e For E1 G 704 without CRC4 default and G 704 with CRC4 The line encoding can be e For T1 AMI no zero suppression mechanism data is encoded in the bipolar Alternate Mark Inversion format or B8ZS Bipolar Eight Zero Suppression For E1 AMI Alternate Mark Inversion standard line coding with no zero code suppression and HDB3 High Density Bipolar 3 code that uses patterns of bipolar violations to replace sequences of 4 zero data bits to maintain ones density on clear channel transmission The signaling can be one of the following T1 CAS In band signaling 24 cha
251. s that help the call reach the desired destination Also for calls transported over a WAN additional delay jitter packet loss and retransmissions can be induced by the network The sum of these delays can exceed the end to end accepted thresholds for example 2 seconds for SIP post dial delay 150 ms for one way delay The standard metrics below can help assess the SIP QoS e Call setup time for the calling part the call setup time measures the time from when an INVITE message is sent until an ACK message is sent Dac Wis START gt INVITE l Er gt lt 2 TART 150 ey ane ee eR Call Setup Time 200 Call Setup Time eee es AcE a STOP Seas an gt 2 TOP Figure 1 SIP call setup time PN 915 2611 01 Rev H June 2014 11 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems For call destinations call setup time measures the time from when the INVITE message is received until the ACK message Is received When the call is authenticated the call setup time includes the authentication time as well Tac Proxy 1 Proxy 2 as INVITE START gt gt l ay 407 See ease ACK el ar ee gt INVITE gt INVITE 100 gt INVITE f lt l i100 gt lt START Call Setup Ti
252. s the quality of transmission As the media information is bandwidth efficient encoded any small loss can affect the overall performance For example a packet loss of 10 greatly affects the decoder ability to build an image this should trigger the call to terminate As a best practice to preserve the user experience within reasonable ratings terminate at a packet loss of 1 The telepresence system should trigger a change to codec bitrate or image resolution in an attempt to lower the active bitrate and packet loss percentage These values represent observations in certain test configurations and the actual parameter values are adjusted according to specific vendor implementations After identifying the sources of network inbound and outbound traffic impairments the system architect should design and implement corrective actions and optimize configuration to overcome them during day to day operation It is important to measure the user QoE especially for interactive media applications Since the early days of telephony there were statics methods to rate the quality of a transmission line As the digital era overtook the analog signal transmissions the methods had to be adjusted for the same need to measure the user quality index The MOS score represents the user point of view metric for QoS over a network as described in the E model defined by ITU in the G 107 document As it is consider a subjective way to evaluate the quality of a transmissio
253. sec VPN Online News Webcast Backup File XFer Business Apps ee VoIP is present in all profiles constituting from 0 5 percent to 15 percent of network utilization Even though only a small part of the overall throughput is used for VoIP the quality of experience QoE for voice calls is critical Users expect the same quality for VoIP calls as for land line service Transporting real time voice over the same network used for all other data traffic presents challenges for service quality Verifying VoIP performance under conditions of high data stress is important to ensure expected results PN 915 2611 01 Rev H June 2014 19 Test Case VoIP Quality of Service in Converged Networks Objective These tests will determine if the DUT supports the desired level of load without degradation of the QoS for VoIP when subject to background traffic The QoS is measured by the call completion rate and quality of voice that is MOS The distribution of traffic follows a subscriber model with a traffic throughput distribution of 85 percent video 22 percent data transfer and 3 percent VoIP Setup Simulated SIP Server Simulated DUT Simulated VoD Server Subscribers Simulated HTTP Server Figure 67 Typical topology for VoIP QoS measurement in the presence of data and video traffic For this example the DUT is a router used as an edge device between home subscribers and the core network The subscribers access the services provid
254. sed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions include further details for configuring the primary parameters of the test and explain additional options that can change the behavior of the test Open the Configuration Template 1 Open the IxLoad GUI 2 Open the VH_002 B2B_ H323v4_NC Basic Call with_RTP rxf configuration template included in Getting Started Templates VoIPH323 File Edit Test Wizards Reporter Options Tools View Help 7 a a BO asse P DUT Terminate a Network a wes Network 1 A Fa Networkt pe E Overview E Networks and Traffic d Mew Traffic Flow El Originate ae S Trafficl Network bl TT Metworkl Figure 44 Overview of the IxLoad configuration sample Configuring the Network Parameters for Network1 Note This network will host 8 000 H 3823 IP Phones Each simulated phone will use a unique IPv4 address and MAC for both H 323 and RTP traffic 1 Click Network1 to display the IP Network Ranges PN 915 2611 01 Rev H June 2014 52 Test Case Determining the Maximum Number of Concurrent Calls 2 Set the following parameters IP Type IPv4 Hosts 8 000 First IP Subnet 20 7 1 7 Mask 8 Increment 0 0 0 1 Gateway 0 0 0 0 Gateway Step Size None 0229 5 p Q Table 14 Summary of Network 1 parameters IP Count Address Mask Increment Gateway Gateway MSS RX Type Increment
255. ses the capacity to sustain the desired volume of VoIP traffic but also a transaction rate for a number of endpoint operations This example demonstrated in detail how to configure IxLoad to maintain a rate of 500 registrations per second for 5 minutes With small changes the same configuration can be used to measure the rate of other type of transactions such as the instant messaging rate PN 915 2611 01 Rev H June 2014 117 Test Case Using VoIP to Measure NAT PAT Performance Test Case Using VoIP to Measure NAT PAT Performance RFCs RFC 1631 May 94 obsolete by RFC3022 The IP Network Address Translator NAT RFC 2663 Aug 99 IP Network Address Translator NAT Terminology and Considerations RFC 3022 Jan 01 Traditional IP Network Address Translator Traditional NAT RFC 4787 Jan 07 Network Address Translation NAT Behavioral Requirements for Unicast UDP Overview Network address translation methods are defined by RFC 3022 which obsoleted RFC 1631 has gained a lot of popularity in today s networks It is a common function found in routers firewalls operating systems and applications Why Use NAT The primary role of NAT is to address the shortage of public IPv4Addresses The 32 bit addressing scheme used by IPv4 allows in theory up to 2 4 294 967 296 IP addresses In practice however the number of real IP public addresses that can be used is actually around 3 2 billion because of a special set of reusable IP ad
256. sing through the communication system versus the reference the signal injected in the system The method requires access to actual audio information in both reference and degraded signals performs level and time alignment to accommodate attenuations and delays process the disturbances and finally applies the cognitive model This is done using signal processing algorithms requiring considerable processing power In packet networks quality of voice measurements can be performed by assessing the packets transmission using the E Model and then generating the metric R Factor As defined by ITU T G 107 R Factor combines a number of values measuring the effect of various impairments some of these are e The effect of coding decoding defined as constants for every codec PN 915 2611 01 Rev H June 2014 87 Test Case Subjective Quality of Voice e Jitter packet loss and delay e The effect of audio signal capture mouth to microphone and reproduction Speaker to ear defined as a constant The E Model doesn t require reference signal information as it does not look at the actual audio content of the degraded signal This method requires far less processing power than PESQ The R Factor method predicts a user satisfaction on a scale from 0 to 100 where 100 is the highest satisfaction A formula is defined for conversions between R Factor and MOS For example a perfect transmission with the codec G 711 has an R Factor of 94 and a MOS of
257. st Case Determining the Capacity of a VoIP to PSTN Gateway Test Case Determining the Capacity of a VoIP to PSTN Gateway Overview Even though VolP is largely adopted there are still a lot of telephony systems in production that use legacy technology The bridge between VoIP and PSTN is done by dedicated gateways that convert the signaling and media between the two environments There is a large variety of gateways depending on type of signaling protocols used in both VoIP and PSTN networks The VoIP to PSTN gateway does the physical and protocol conversion of both signaling and media traffic The basic SIP to ISDN call flow is as follows UAS Gateway PBX l INVITE lt gt Disconnect BYE Se 6 eee Peso sens l jJRelease l GE gt Se Release Complete l nee gt Figure 145 Basic SIP to ISDN call flow Objective The goal of this test methodology is to determine the capacity represented in the number of concurrent calls that can be handle by a VOIP to PSTN gateway IxLoad simulates SIP User Agents on Ethernet interface and PSTN PBX on TDM interface The detailed steps below will show how to configure a test in IxLoad to originate calls from SIP and terminate on E1 PRI to test a getaway E1 is a format of Time Division Multiplex TDM technique providing 2 048 Mbps communication link divided into 32 time slots of 64 kbps each E1 is used in most parts of the world while a different
258. stain Time or Packets Lost and compare the RTP Packets Sent with RTP has lower values than the Outgoing Packets Received This issue may be caused by a large RTP Throughput number of packets being dropped or by calls without RTP packets Endpoints connected in calls without receiving any media are counted using the Calls without RTP packets Statistic All calls were connected but the Verify that requests to disconnect the calls are not received expected maximum throughput is not from the DUT during the Talk Time reached The incoming throughput and outgoing Verify the Codec Distribution and RTP Packet Distribution throughput values are lower or higher Statistics these may be used to confirm that the calls than expected negotiated the desired audio video CODEC Problems may be fixed by correcting the DUT configuration or by setting the same CODEC on both calling and receiving activities Conclusions This test methodology provided a guideline for determining the maximum number of voice or video calls that can be established through a device with NAT functionality enabled by emulating clients in the private network and the server in the public network The mapping of IP addresses and port numbers used in the private network correlated with the traffic type traffic volume number of concurrent streams or combination of voice and data traffic can lead to QoS degradation or the absence of media PN 915 2611 01 Rev H June 2014 154 Te
259. stant call rate Setup In this example a pair of Acceleron NP ports are used Calls will send voice traffic bi directionally and will have a call hold time conversation time determined by the test objective The test will run for five minutes The configuration simulates a SIP network Network1 with 8 000 IP phones and a secondary SIP network Network2 which includes one SIP Proxy Server and one Media Server with 8 000 phones behind it Enterprise 1 B1B SXX INNN 5 ee 4 a P mizi xe 5 ne eee Network 1 MA Enterprise 2 i e 919 5XX 1NNN Network 2 Figure 9 Test Topology Ixia generates and receives the traffic passing through the WAN PN 915 2611 01 Rev H June 2014 16 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Step by Step Instructions The final IxLoad configuration as a result of these steps is provided on the blackbook ixiacom com Web site IxLoad 5 10 Voice SIP Call Setup Rate crf To import a Compressed Repository File crf in IxLoad use the command Import under the File menu The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test Open the Configuration Template 1 Open the IxLoad GUI 2 Open the VS _ 002 B2B_SIPv4 MakeCall ReceiveCall EndCall with RTP 33s rxf configuration template included in Getting
260. stribution statistics Statistic Name VEIO Questions Packets with Delay Variation orre See Jitter up to 3 ms ase Jitter up to 5 ms aree Jitter up to 10 ms etl Jitter up to 20 ms 1 Assuming Jitter was reported what is the distribution of the Delay Variation Jitter values Packets with Delay Variation Jitter up to 40 ms Packets with Delay Variation Jitter over 40 ms Table 52 Distribution of RTP Consecutive Lost Packets Statistic Name Name VEUT E eT io a Packet Sequence Three Packet Sequences Ten Packet Sequences or More Packet Sequence C We L EE an 1 Assuming that packet loss was reported ga me T eas ae what is the distribution of the lost RTP or Five Packet Sequences packets PN 915 2611 01 Rev H June 2014 153 Test Case Using VoIP to Measure NAT PAT Performance Table 53 RTP Streams Statistic Name VEE O Questions Concurrent RTP Streams Assuming that packet loss was reported what is the distribution of the lost RTP Concurrent RTP Streams ae Packets max incoming RTP packets Are any calls with RTP wt 1 Does this number match the number of incoming RTP packets Calls Connected 2 Troubleshooting and Diagnostics The following table summarizes some of the common issues that may be encountered when running a call rate test Table 54 Troubleshooting Solution The Incoming RTP throughput is not Check the reported RTP Packets Lost RTP Consecutive constant during the Su
261. t supported by IxLoad Secure RTP TOS DSCP marking for H 323 amp RTP a fo individual settings for H 323 amp RTP DUT Test Variables Table 27 Parameter Name Current Value Additional Options IP version IP version PVA PVG Transport Protocol SIP UDP SIP TCP SIP TLS Conclusions This example demonstrated how to configure IxLoad to maintain 8 000 calls active for 220 seconds This configuration allows you to measure the maximum number of simultaneously active calls Guidance on how to assess the QoS for media and signaling is also included PN 915 2611 01 Rev H June 2014 74 Test Case VoIP Quality of Service in Converged Networks Test Case VoIP Quality of Service in Converged Networks Overview The migration to packetized voice VoIP is driven by the desire to use existing data networks for application traffic video and voice This means that effective VoIP performance testing cannot be done in isolation because few networks run VoIP exclusively Services of all types that use a range of protocols are seen in modern multiplay networks including Data HTTP HTTPS FTP E mail Voice over IP SIP MGCP RTP IPTV RTSP IGMP Peer to peer BitTorrent eDonkey Gnutella Infrastructure DHCP DNS RADIUS Security SSL TLS IPsec The table below illustrates typical network usage for different user profiles Table 28 Traffic distribution profiles Corporate Telecommuter GenY E Mai FTP P
262. t an existing unlinked activity PSTNDigitalPeer at step3 Close the window by clicking Yes 5 Click the PSTNDigitalPeer1 activity PN 915 2611 01 Rev H June 2014 163 Test Case Determining the Capacity of a VoIP to PSTN Gateway 6 Inthe Scenario tab add and connect the T1 E1 script objects Make Call Voice Session and End Call Add triggers to between the SIPMakeCall activity and PSTNDigitalPeert activity Traffice PSTNDigitalPeer1 PSTNDigital Peer Execution Settings Dial Plan Audio D ae id Ba couo 4 SM x e Reche E ee ee a e H Tiel He Add Channel ERemove Ee Compact l Note Zoom 100 Scenario Channel 0 Activity SIPMakeCall Traffic Traffic Hetwork1 Column Originate Link YoiceLink1 Digital only by Make Call fat Receive Call Voice che Talk ee Listen 125 Generate DTMF 123 Detect DTMF 123 Generate MF 129 Detect MF fl Generate Tone Wait For Tone ke seen Path Confirmation Procedure Librar Figure 154 SIP PSTN test scenario Edit the Dial Plan for PSTNDigitalPeer1 1 Click the Dial Plan tab 2 Edit the Source Phone Number Specify field to 717444 0001 0030 this is the same sequence defined for the SIP Make Call activity as Destination Phone Numbers Note Depending on the distribution group defined on the DUT IxLoad may be configured for other matching schemes of incoming calls DNIS to receive ANI to receive
263. t the network parameters as shown in the following table Table 1 Summary of Network2 parameters Parameter IP Range 1 will be used for IP Range 2 will be used for RTP SIP IP Type 20 1 100 1 20 1 200 1 Mask B Note To simulate a SIP trunk which receives and generates both SIP and RTP traffic using a common IP address set a single network range instead of two that is skip the preceding step 2 Stack 2 gt filte 2 f Z TCP 2 Settings 2 GratARP 2 gt DNS 2 Ethernet 2 Enabled Gateway Increment Gateway Increment Mode bled IP R2 Unconfigured IPy4 20 1 100 1 a 0 0 0 1 1 0 0 0 0 0 0 0 0 Increment every subnet 1460 IF R3 Unconfigured IPw4 20 1 200 1 6 0 0 0 1 1 0 0 0 0 0 0 0 0 Increment every subnet Figure 13 Configuration example for Network2 Configuring Traffic 1 for Network1 The traffic element includes options that allow one or more IP ranges to be associated with one or more activities Similarly every VoIP activity allows signaling and media to use distinct ranges The default settings associate the same IP address with SIP and RTP traffic Because Network1 simulates SIP IP phones that generate SIP and RIP traffic using the same IP source we can use the default settings Select Traffic1 from Network1 1 For Network Range IP R1 included in Network1 Error Reference source not found verify that a Group uses the Consecutive IPs distribution rule PN 915 2611 01 Rev H
264. t time 20 ms 160 bytes per Frame and G11 p Law a Custom Codec ITU T G 711 is a standard to represent bit compressed pulse code modulation PCM samples For signals of voice Frequencies sampled at the rate of 8000 samples second 6 711 encoder creates a 64 kbps bitstream Figure 27 Payload type Incoming payload type Outgoing payload type Will be used For RTP header payload type when sending packets and viceversa Please note khat the outgoing payload type of a sender must be equal to the incoming payload type of a receiver lt gt WARNING The outgoing payload type number Configuring CODECs settings for the Make_Call activity Note If the required codec is not natively supported by IxLoad the Custom Codec can be used this feature allows playing of any RTP stream provided as a traffic capture Configuring RTP Settings for the Make_Call Activity 1 Click the RTP tab 2 Ensure that you have selected the Enable Hw Acceleration check box to allow 8 000 RIP streams per Acceleron XP port Hardware acceleration _ RTCR Calculate advanced statistics AudioVideo port 10000 Per Stream Statistics MDI Statistics _ Non blocking execution Figure 28 PN 915 2611 01 Rev H June 2014 Verify all settings Restore defaults Configuring RTP settings for the Make_Call activity 29 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Configuring Audio Se
265. tatistics highlighting the pass fail result based on call flow execution Statistic Name Questions 1 Are the Successful Loops and Total Loops values equal Successful Loops 2 Have any Failed Loops Aborted Loops or Failed Loops Warning Loops been reported PEELO Note failed aborted and warning loops highlights Warning Loops failures at the scenario level Total Loops Troubleshooting and Diagnostics The following table summarizes some of the common issues that may be encountered when running a call rate test Table 61 Issue Troubleshooting Solution The number of Check the SIP event viewer a pattern in the phone numbers failing to concurrent calls is not connect calls is an indication of miss configuration of some phones wrong sustained for the destination phone number entire test duration Check the SIP Retransmissions counter Call Setup Time and End Call Time measurements A high number of retransmissions is an indication that the SUT cannot maintain the load generated This leads to larger call setup and teardown times which affects the number of users available to place new calls PN 915 2611 01 Rev H June 2014 171 Test Case Determining the Capacity of a VoIP to PSTN Gateway Test Variables Table 62 Parameter Name Current Additional Options EU Up to 480 concurrent calls using a single adapter card with two eight spans E1 T1 modules 2 x 8 x 30 More than 480 concurrent calls using more than one
266. te for VoIP Protocols Network Traffic Mapping Objective Type Objective Value Iteration Time Total Time E f New Traffic Flow B amp Activity Links Jh VoIPLink2 Loops Initiated Per Second D Timeline1 000 06 23 000 06 23 7 YoIPSipPeer 1 Public Loops Initiated Per Second 500 Timeline1 000 06 23 000 06 23 A gt User_Agents Private Loops Initiated Per Second 500 Timeline 1 000 06 23 000 06 23 Timeline Custom Parameters Timeline Name Timeline C Advanced Ramp Up Type Channels Interval Ramp Up Value Scaling type Auto Ramp Up Interval s00 Ramp Up Time 0000 00 43 Sustain Time 0000 05 00 400 Ramp Down Value 300 Ramp Down Time 0000 00 40 ew VoIPSipPeer 1 Iteration Time 0000 06 23 200 Iterations 100 Time to First Iteration 0000 00 00 Iterations 0 Time Between Iterations 0 00 00 0 00 41 0 01 22 0 02 03 0 02 44 0 03 25 0 04 06 0 04 47 0 05 28 0 06 23 Figure 103 Test objective and Timeline For this test the test objective is 500 loops per second validating whether a rate of 500 registrations per second can be maintained 11 Map the ports a pair of Acceleron ports is enough for this test Running the Test 1 Click Run to start the test execution 2 IxLoad will automatically display Statistic Views after execution starts Results Analysis The following questions provide guidelines on how to recognize problems during the test or in post analysis 1 To determine if the
267. ted by the service provider Let s assume Alice wants to call her friend Bob Before placing the call her phone must register by sending a REGISTER request to REGISTRAR s public IP 192 16 148 244 5060 1 REGISTER 2 REGISTER REGISTRAR amp PROXY fa A emai 193 16 148 244 5060 Alice 3 200 OK 75 83 202 16 Figure 120 SIP Registration fails when attempted behind a NAT NAPT device The 1 REGISTER request initiated by Alice s phone has the following content SRC IP SRCPORT DESTIP DEST PORT 192 168 1 100 193 16 148 244 REGISTER sip myserviceprovider com SIP 2 0 Via SIP 2 0 UDP 192 168 1 100 5060 From Alice lt 51p 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt sip 8184443118 myserviceprovider com gt Contact Alice lt s1p 8184443118 192 168 1 100 5060 gt Content Length 0 Figure 121 IP header and IP payload in 1 REGISTER On receipt of the REGISTER request the NAT device located at the edge of the enterprise network changes the source IP address and port number from the IP header of the REGISTER packet leaving the destination IP address and port unchanged The payload is not modified The first available port number is selected for the port mapping let us assume 1025 The REGISTER request has the following structure PN 915 2611 01 Rev H June 2014 132 Test Case Using VoIP to Measure NAT PAT Performance 5060 REGISTER sip myserviceprovider com SIP 2 0 Via S
268. tery life e VoLGA Voice over LTE via GAN The VoLGA standard was based on the existing 3GPP Generic Access Network GAN standard and the goal was to enable LTE users to receive a consistent set of voice SMS and other circuit switched services as they transition between GSM UMTS and LTE access networks e Voice over LTE VoLTE initially called One Voice The Voice over LTE VoLTE aims for providing voice over an LTE system utilizes IMS enabling it to become part of a rich media solution One additional approach which is not initiated by operators is the usage of Over the top OTT content services using applications like Skype and Google Talk to provide LTE voice service However handing the LTE voice service over completely to the OTT actors is expected to not receive too much support in the telecom industry while the voice call service is and will still be the main revenue source for the mobile operators The typical topology for VoLTE is shown in the VoLTE Topology The SIP registration and call control messages are sent from the User Endpoint UE over the default bearer in EPC to the Proxy Call Session Control Function P CSCF the entry point in the IMS domain In some networks an Session Border Controller SBC is used for this function The Serving Call Session Control Function S CSCF is the central node of the signaling plane It is a SIP server that communicates to the Home Subscriber Server to download the users profile
269. the endpoint and Registrar SIP server are separated by a NAT device the register message also has the role of opening a firewall pinhole from the server to the endpoint Enterprise NATINAPT REGISTRAR amp PROXY e a a a a ao ao co cD aD ee eee eee eee eee aD ee eee aD aD SD a aD SD a a a a a a a a al al al al a Figure 92 SIP endpoints behind a firewall NAT devices will discard any messages originating from a SIP proxy server located in the public network unless an initial message from an endpoint has been sent to the Registrar and Proxy Even after the pinhole has been opened it only stays open for a limited time To ensure that the endpoint is reachable at all times it must retransmit messages to the SIP servers through the NAT device The interval between retransmits must be small enough to keep the pinhole open typically 30 seconds If there are a large number of endpoints this can add up to a great deal of traffic more than that associated with call setup For example if there are 1 million users behind a NAT device who register every 30 seconds without authentication only 2 messages are required there will be 33 333 registration requests per second For the same number of users with a nominal call rate of 2 calls per hour there will be 556 calls per second This equates to 133 Mbps of bandwidth for registration versus 11 Mbps for call control With a transaction rate 30 times larger than call control and wit
270. the maximum capacity is determined a system characterization can be made using longer execution time for example minimum 72 hours to confirm that the system maintains the QoS in the expected range To plot the performance characteristic of the DUT at least 10 data points are recommended for the chart 5 percent 10 percent 25 percent 40 percent 50 percent 65 percent 75 percent 85 percent 95 percent and 100 percent of the maximum supported rate PN 915 2611 01 Rev H June 2014 49 Test Case Determining the Maximum Number of Concurrent Calls The performance characterization of the DUT should continue by changing test variables that influence the performance of the DUT Those test variables include e H 323 call setup mode normal call fast start tunneling H 245 in parallel with fast start e Negotiated CODEC e IP version IPv4 IPv6 e IP and port mapping for signaling and media 1 1 1 n e Call duration e Number of CODECs supported per user lf the media traverses the DUT the negotiated CODEC type must also be considered The CODEC used is even more important when the DUT acts as a transcoder for example converts the voice from G 711 to G 729 Objective The test objective is to connect 8 000 calls keeping them active for 5 minutes This simulation will use 8 000 users with every phone generating a single call Using a call rate of 100 cps results in all the calls being active after 80 seconds 8 000 calls 100 cps
271. the public network This is possible by configuring the NAT device as a gateway to the public network However the NAT function cannot by itself support all applications transparently and often must co exist with application level gateways ALGs for this reason Except for ALGs NAT devices do not examine or modify the payload of the packet For this reason NAT devices can often cause difficulties Some situations where traditional NAT will not work are when an application payload itself includes an IP address or when end to end security is needed IPsec techniques which are intended to preserve the endpoint addresses of an IP packet will not work with traditional NAT because protocols such as AH and ESP protect the contents of the IP headers including the source and destination addresses from modification Yet NAT s fundamental role is to alter the addresses in the IP header of a packet PN 915 2611 01 Rev H June 2014 120 Test Case Using VoIP to Measure NAT PAT Performance IPSec Disabled IPSec Tunnel Mode and AH New IP Header AH Header IPSec Tunnel Mode and ESP New IP Header ESP Header Figure 106 IPsec AH and ESP headers in Tunnel Mode NAT Concept The example from the figure shown below explains the NAT concept using a simple private network consisting in two hosts connected to the public network using the customer edge CE router which is NAT enabled The hosts use private IP addresses from class C 192 168 1 10
272. the voice traffic Redundancy measures the ability to continue operation in the event of a failure Performance measures the maximum rate that a system or device under test DUT can sustain For IP telephony systems the key performance measurements are the maximum call setup rate and the maximum registration rate The performance of systems without session intelligence for example a stateless proxy server can be additionally measured using metrics such as maximum transaction rates or maximum messages per second In addition to performance measurements capacity tests help in determining the maximum number of active calls that can simultaneously be active on a DUT While the presence of media has an immediate impact on the performance and capacity of a DUT that forwards media packets between two hops another system may not be impacted For example because only PN 915 2611 01 Rev H June 2014 3 Voice over IP session internet protocol SIP messages can traverse the DUT While it is recommended that media traffic be included with any call testing media may or may not be generate during the Call Hold Time as part of the test The decision needs to be taken based on the logical components included in the system under test SUT As an example a session border controller SBC must be tested by generating media after calls are established while testing of a SIP proxy server will not necessarily require media While media traffic may not aff
273. this case the number of IP addresses should be equal with the number of ports used for servers simulation Networks and Traffic Mew Traffic Flow Originate al a Homesubscribers Ea HomeSubscribers i E v rrvvideo a iw IPT yServer a HTTFClient N E HTTPServer Ee i SIPClient a Subscriber 1 IPT ideo Command Editor Start Start l EF Get 1 Ri sie Makecall Corn i Think 2 e Error Gh Voice Session Stop OK Error M sir Endcall Initiate Error Stop Figure 68 IxLoad configuration for multiplay testing 2 Inthe Servers network add an IPTV Video Server activity In this example the activity has been renamed to PTVServer In the Video Config page set the Stream Count parameter to 40 and Duration to 300 seconds In the Advanced Options change the UDP Port Range to 70000 20000 the same IP address is used for Video and VoIP traffic so the UDP port domain is shared between the IPTV and VoIP activities Keep the rest of the parameters at their default values The test is now configured for unicast video streaming the voice QoS should be tested also with multicast video traffic 3 Inthe Server network add an HTTP Server activity In this example the activity has been renamed to HI TPServer Leave the default parameters unchanged PN 915 2611 01 Rev H June 2014 78 Test Case VoIP Quality of Service in Converged Networks 4 Inthe Hom
274. those that you type PN 915 2611 01 Rev H June 2014 Vil Voice over IP Dear Reader Ixia s Black Books include a number of IP and wireless test methodologies that will help you become familiar with new technologies and the key testing issues associated with them The Black Books can be considered primers on technology and testing They include test methodologies that can be used to verify device and system functionality and performance The methodologies are universally applicable to any test equipment Step by step instructions using Ixia s test platform and applications are used to demonstrate the test methodology This tenth edition of the black books includes twenty two volumes covering key technologies and test methodologies Volume 1 Higher Speed Ethernet Volume 12 Pv6 Transition Technologies Volume 2 QoS Validation Volume 13 Video over IP Volume 3 Advanced MPLS Volume 14 Network Security Volume 4 LTE Evolved Packet Core Volume 15 MPLS TP Volume 5 Application Delivery Volume 16 Ultra Low Latency ULL Testing Volume 6 Voice over IP Volume 17 Impairments Volume 7 Converged Data Center Volume 18 LTE Access Volume 8 Test Automation Volume 19 802 11ac Wi Fi Benchmarking Volume 9 Converged Network Adapters Volume 20 SDN OpenFlow Volume 10 Carrier Ethernet Volume 21 Network Convergence Testing Volume 11 Ethernet Synchronization Volume 22 Testing
275. time Media Flows Call Termination Call Termination SS Stopping Real time Media e Stopping Real time Media Figure 205 Multipoint telepresence simplified call flow Typical Cisco s implementation for telepresence call has three cameras and three large high definition displays to enhance the user experience and several microphones in the room to capture the audio signal Each stream is coded individual and transmitted to the other side using the negotiated codecs from SDP The codec type and codec settings will dictate the necessary bandwidth that RTP data generates For better interaction there is an additional presentation stream that provides the platform to share content These aspects are to be considered when designing converged networks since the maximum capacity of the call can affect the inbound or outbound points of the network The following table provides an example of quality requirements in terms of bandwidth for H 264 video codec and AAC LD audio codec PN 915 2611 01 Rev H June 2014 217 Test Case Measuring Quality of Experience for Multimedia VoIP Calls Table 76 Telepresence single endpoint bandwidth requirements Video Motion Bandwidth Bandwidth Interactive Compatibility Aggregated Quality quality per video per channels media Bandwidth stream microphone channels required 1080p 4Mbits 64Kbits 564Kbits 832Kbits 13 6Mbits 1080p 3 5Mbits 64Kbits 564Kbits 832Kbits 12Mbits To understand the call flow that is about to take pla
276. tion and resource reservation and manages traffic functions like call routing operations This layer gathers information from the user interface and allows users to reserve and create meeting rooms send invitations or remove users from a conference Control Plane Layer This layer is responsible for the signal stack The most common signal is the Session Initiation Protocol and H 323 This offers the functionality to interact with the endpoints in an attempt to establish the call or conference for inbound and outbound and also negotiate the session parameters that all complies to Media Plane Layer This has the function to mix and deliver the media streams of video and audio payload to and from the joining endpoints The main function of this layer is to control the function and delivery of media by management of the Real Time Transport PN 915 2611 01 Rev H June 2014 214 Test Case Measuring Quality of Experience for Multimedia VolP Calls Protocol and Real Ti me Transport Control Protocol suites It assists in process of negotiation of the payload type codec in use video frame rate video resolution and additional media channel parameters that the teleconference can have From topology deployment point of view the telepresence application can be categorized as point to point call referred to as business to business or intra campus conferencing This is also called the multipoint videoconferencing or multi site meetings These resources a
277. tion example for the Audio Settings page PN 915 2611 01 Rev H June 2014 60 Test Case Determining the Maximum Number of Concurrent Calls Configuring the Execution Settings for the Receive_Call Activity 1 Click the Receive_Call activity 2 Click the Execution tab 3 Verify that the script will be executed a single time during the Sustain Time the Run for to A number of loops is set to 7 4 Verify that the Graceful Ramp down check box is selected Note This option forces the users to hang up the call when the ramp down request is received in the middle of the call Traffic2 Receive_Call VoIPH323 Peer Scenario fection ial Plan 323 Terminal Capaity Codecs RTP aud Video Other Fun For Loop delays C The entire test duration Before ist loop 0 ae A number of loops 1 Between loops O ms Phone Number Mapping Rules For H323 Graceful Rampdown Phone Number Use consecutive values per acti Graceful Ramp down Figure 55 Configuration Example for Execution Settings page of VolPPeer2 5 Under Phone Number Mapping Rules for H 323 set Phone Number to Use consecutive values per port Configuring the Dial Plan for the Receive_Call Activity in Network2 1 Click the Dial Plan configuration page of Receive_Call 2 Set the Source Phone numbers by selecting the User defined check box and enter the sequence 919501 1000 PN 915 2611 01 Rev H June 2014 61 Test Case Determining the Maximum Number of
278. tions provide guidelines on how to recognize specific problems during or at the end of the test execution 1 Has the test objective been achieved Check the Call Rates view Statistic Name Calls Attempted per Second Calls Connected per Second PN 915 2611 01 Rev H Table 2 Call Rate statistics Questions Have the calls been attempted continuously at a constant call rate during the Sustain Time How do the Calls Attempted rate and the Calls Connected rate compare to each other June 2014 39 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems 2 Have any call failures been reported Check the Calls view Table 3 Call statistics Statistic Name Value Questions Calls Attempted 1 Have any call attempts failed Calls Connected Compare Calls Received a Calls Attempted and Calls Received with Calls Answered b Calls Attempted and Calls Connected End Calls Initiated Have any attempts to hang up the call End Calls Received failed Compare a End Calls Initiated and End Calls Received with b Calls Attempted and End Calls Received Have all the calls attempted ended End Calls Completed successfully Compare 2 Calls Connected with End Calls Completed 3 Why do we need to compare End Calls Completed with twice the number of calls connected 3 Have any scenario loop failures been reported Check the Loops statistics view Table 4 Statistics highlig
279. tory File crf in IxLoad use the command Import under the File menu Follow the wizard to save on the local drive the included files The step by step instructions highlight how to set the essential parameters of this configuration and explain additional options which may be used to change the behavior of the test or create new test scenarios PN 915 2611 01 Rev H June 2014 23 Test Case Telephony Denial of Service Setting the Network Parameters 1 Start IxLoad application i E E P Add Net Ade Add Remove Add ImpairNet Traffic Activity DUT App lix Z Edit Impairment Networks and Traffic Scenario1 Originate Figure 223 Adding a new originating NetTraffic Table 86 Sample network configuration Parameter Value IP version IPv4 Mask 24 Count 100 PN 915 2611 01 Rev H June 2014 238 2 Add anew VoIPSIP Peer activity by clicking the button As necessary expand the Test Case Telephony Denial of Service Voice section from the drop down control window Originate PN 915 2611 01 Rev H C Most Recently Used OG e e e Figure 224 Adding a new VolPSIP Peer activity June 2014 DNS HTTP LDAP NFS SME Attack Data Storage Video Voice Bulk MGCP Bulk SIP H248 MGC H248 MGW MGCP CA MGCP GW VoIPH323 239 Test Case Telephony Denial of Service Click the VoIPSIP Peer activity and select the Scenario control tab This contains the controls for call flo
280. tributes of the system and the participants speaking and listening preferences The ITU T P 800 specification defines methods for subjective determination of transmission quality These methods utilize a large number of human subjects who listen to sentences read aloud by professional male and female speakers and transmitted over the telephony system The listeners rate the quality of the audio signal Individual results are averaged into a MOS that provides a numerical indication of the quality of transmission in the range of 1 to 5 Table 36 Mean Opinion Score Quality Excellent Good Fair Poor Bad PESQ perceptual evaluation of speech quality is a mechanism that measures the quality of speech in an automated way defined by ITU T standard P 862 PESQ is an objective measurement method that predicts the results of subjective listening tests The PESQ algorithm produces results analogues with the subjective MOS standard A mapping between PESQ results and MOS was defined after the release of the P 862 recommendation PESQ LQ PESQ Listening Quality is defined in ITU T Rec P 862 1 and improves the correlation with subjective test results at the high and low ends of the scale Ix_Load measures both values PESQ LQ and PESQ LE Listening Effort PESQ is a full reference method designed for end to end quality of voice assessment using a psycho acoustic and cognitive model PESQ analyses the degraded audio signal the signal after pas
281. ts Analyzer Originate alll Test Overview Networks and Traffic F New Traffic Flow ag HI Originate Y iaraonenora Accessing network configuration details Figure 212 PN 915 2611 01 Rev H June 2014 225 Test Case Measuring Quality of Experience for Multimedia VolP Calls 6 Configure the necessary IP addresses subnet mask IP count for simulation and default gateway details NOTE e Several IP address ranges can be configured in parallel to emulate large networks or different office branches by accessing the button e xLoad offers support for IPv4 and IPv6 Before test execution verify in the release notes or application datasheet if the configured protocol has adequate support for that IP version Network1 s gt Settings GratARP 1 S gt DNS 2 gt fFilter 1 p TCP 2 Pa MAC vVLAN 1 A Ethernet 1 trated name status pType Adress Mask increment count Gateway IP Ri Unconfiqured IPv4 192 1563 21 15 16 0 0 0 1 5 192 168 254 250 PSCC a Figure 213 Changing the network connectivity details Table 77 Template network configuration Endpoint IP Address IP Mask IP Count Default Gateway 23 1 1 16 bits 0 0 0 0 2 3 100 1 16 bits 0 0 0 0 7 Adjust the network configuration to access the application endpoint emulation on the Originate side 8 Click the VolPSipPeer1 peer to access the endpoint s settings Originate Fig
282. ttings for the Make_Call Activity 1 Click the Audio tab Ensure that you have selected the Enable audio on this activity check box 2 3 Specify the clip to be played 4 Specify the duration of the play this will determine the call duration 5 Select the MOS and One Way Delay check boxes if these metrics are of interest for the test that you want to configure Enable audio on this activity if unchecked all audio script Functions will be SKIPPED Play Settings _ Enable jitter buffer Clip US O42 ware Format PCM Duration 32785 ms Size 524556 bytes Output level 20 dBm G Play for clip duration or TalkTime all objectives except Channels C Play For _ Perform Qov _ Type OF Service TOSIDSCP Calculate One Way Delay _ Generate silence Figure 29 Audio Settings Note The Clip and Play Duration settings may be overwritten by the Voice Session functions In this example the settings at the activity level from this tab are used because the Voice Session functions have the Overwrite check box cleared Configuring Execution Settings for the Receive_Call Activity 1 Click the Receive_Call activity 2 Click the Execution Settings page 3 Set the activity to run the script continuously during the Sustain Time using the Run for the entire test duration option This will ensure that the CPS rate will be constant for the entire Sustain Time Note The Run for parameter accepts a fixed number of loops as
283. ty of applications to work through existing NAT infrastructures Using STUN applications can discover the presence and the types of NATs and firewalls between them and the public Internet It can also be used to determine the public IP addresses and ports allocated by NAT STUN works with many existing NATs and does not require any special behavior from them To determine the IP port mapping information STUN uses an external STUN server STUN is not a NAT traversal solution by itself but it can be used as a tool in the NAT context For example a VoIP client can use STUN to discover whether it is behind a NAT determine the NAT type discover the public IP address and port number on the outermost NAT and then utilize that IP address and port within its protocols A system using STUN will include one or more STUN clients located in the private network and one STUN server located in the public network The following example assumes that a STUN client is located on a SIP endpoint in the private network and uses an external STUN server to discover the IP and port mapping information as well as opening the SIP signaling port 5060 The figure below illustrates the first step in the communication between a STUN client and server STUN client 192 168 1 102 5060 sends the STUN Binding Request to the public STUN server 209 132 176 30 3478 r7 Private Network Internal Public Network External i NAT NAPT 192 168 1 102 24 SRC IP
284. ulated user agents m Authentication UAC Override registrar IP PORT User name User160 00000000 Password Passi60 00000000 Construction of SIP messages T Override default contact settings Edit Contact rt Type Of Service Override default destination domain name or host port TOS DSCP Best Effort 0x00 Domain name or Host Port Use Tel URI scheme for Source Use Tel URI scheme for Destination Transfer address Edit Verify all settings Restore defaults Figure 101 SIP settings for emulated endpoints 8 Define the SIP registrar IP address To register a SIP device the IP address of the registrar server must be known This can be specified in the SIP settings property page for the VolPPeerSIP activity Refer to above figure On the emulated endpoints activity the Originate network traffic select the following check boxes e Use external server Place the IP address of the Registrar server in the Server address In more complex networks this field will contain the IP address of the proxy or session border controller serving the emulated endpoint The Outbound Proxy check box should be selected if you want all the SIP messages to be sent through this server this is a typical case for NAT topologies e Registrar Server By default the IP address specified in the Server address field is used as the IP address of the Registrar server When the Proxy and Registrar servers have different IP address
285. umber of Concurrent Calls 5 For both Voice Session script objects included in the script do the following a Double click the Voice Session script object Its properties open the Talk Parameters tab appears b Clear the Overwrite playback activity settings check box c Click the Listen Parameters tab d Clear the Overwrite playback activity settings check box e Click OK to close the properties page f Repeat Steps a to e for the second RTP Talk script object Yoice Session Properties Zol Yoice Session Properties _ oO Talk Parameters Listen Parameters Advanced Settings Output Settings Listen Parameters Advanced Settings Output Settings Delay Before Execution Listen Settings Static Expression gt f Listen duration 23000 ims F 0 m f Use Talk lime for all objectives except Channels Talk Parameters Perhorn Gov measurements Clip Y Output level on qr Output Volume 20 dBm U 0 7745 Y A 600 ohm Clip L5 042 wav T Output level 20 dm PCM 8000 Hz Mono Size 524556 bytes Time 32785 ms t Flay fi timels i Repeat continuous for fico Sec f Use Talk Time for all objectives except Channels f Use Global settings Plays one or more WAYE Files and listens for audio RTP at the same time The re Plays one or more WAYE Files and listens For audio RTP at the same time The r wave Files can have the Following characteristics Wav
286. une 2014 141 Test Case Using VoIP to Measure NAT PAT Performance SBC s solution for non routable media flows To address this issue the signaling border element SBE component of the SBC implements the SIP B2BUA functionality for the signaling protocol which allows the call between the calling and called parties to be separated in two distinct conversations one between the caller and the SBC and one between SBC and called party After the media is established the data border element DBE component of the SBC acts as an RTP proxy between the caller and called party The following figure illustrates the path of the INVITE SDP request traveling from Alice s phone to the proxy server and the path of the 200 OK SDP response The IP addresses and port numbers for the signaling messages are highlighted at the top The media IP addresses and media port numbers are highlighted at the bottom The SDP portion used to describe the media IP address and port number used in the INVITE and 200 OK response messages is also highlighted at the bottom The SDP parameters used to inform the remote party of the IP address and port number that are used to generate and receive media are specified under the c line which contains the media IP and the m line which contains the media port number SRC 192 168 1 100 5060 REGISTRAR amp PROXY SRC 172 16 100 9 5060 16 100 2 DST 193 16 148 244 5060 SRC 75 83 202 16 5060 par 172 16 100 2 Sae0 172
287. up1 to MediaGateway and check that the group has Consecutive IPs rule set right click the IP Group Edit Group b Close the Modify distribution group by clicking OK 6 Move the Network Range IP R3 under the MediaGateway distribution group select the Network Range and click the arrows to move it up or down Activities amp Endpoints Network Ranges By Port Distribution Group Receive_Call El IP 2 SIPProxy Consecutive IPs Network Range IF R2 in Network 0 1 100 14 1 K EI E sjaja sjaja Network Range IP R3 in Network 20 1 200 1 1 MediaGateway Consecutive IPs fiil m m Figure 18 Network Range 7 The MediaGateway group has the Consecutive IPs rule set PN 915 2611 01 Rev H June 2014 21 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems a The network range included under MediaGateway has only the RTP check box selected Note The traffic toolbar _ allows you to create and delete Port Distribution groups and to select the IP distribution rule within a group Consecutive Ranges Consecutive IPs and Round Robin You can move a Network Range from one group to another using the Up Down buttons available in the same toolbar In this example the group labels are set to S IPProxy and MediaGateway for clarity Configuring a Test Scenario for the Calls Attempted per Second Objective The media duration configured inside Scenario Editor plays a critical role in achievi
288. ure 214 Accessing application endpoint settings PN 915 2611 01 Rev H June 2014 226 Test Case Measuring Quality of Experience for Multimedia VoIP Calls 9 Select the Audio tab and verify the current configuration To measure the quality of soeech for the emulated traffic enable Perform MOS and Calculate One Way Delay options During test execution these will provide real time statistics for the emulated endpoints as well as aggregated data for the entire call rating NOTE If the option to calculate MOS is enabled the IxLoad user interface enables the option to measure one way delay Scenario Execution Enable audio on this activity if unchecked all audio script functions will be SKIPPED Play Settings _ Enable jitter buffer Clip UK 31 Telepresence438 mp4 Format AAC LD Duration 40020 ms Size 320941 bytes Output level 20 dBm _ Play for dip duration or TalkTime all objectives except Channels Play for 30 seconds C Perform Qov _ Type Of Service TOS DSCP Perform MOS Calculate One Way Delay Generate silence Null data encoded Comfort noise Verify all settings Restore defaults Figure 215 Enabling the voice quality measurement NOTE Use the template configuration that has the actual telepresence call of an audio stream sent for 30 seconds The play duration value can be adjusted to better suit the scope and duration of the test 10 Select the Video configuration tab and verify the cur
289. ure 218 Enabling the voice quality measurement 11 Repeat steps 5 to 9 for the Terminate side NetTraffics on the VolPSipPeer2 telepresence activity endpoint The application simulation is now ready for execution using the default scenario activity emulation for triple screens and HD1080p streams 12 Select the Timeline and Objective option from the menu tree and set the required test objective For initial testing of network convergence it is suggested to set 1 channel objective and work the way up to measure the maximum performance of the SUT Network Traffic Mapping Objective Type Objective Value Iteration Time F New Traffic Flow Activity Links Git VoiceLinkt 00 05 11 0000 05 11 Wi volPSipPeer1 Network1 Channels 1 000 05 11 00 05 11 ia VoIPSipPeer 2 Network Channels 1 000 05 11 O00 05 11 Figure 219 Adjusting the test objective The test is ready now for execution It is necessary to assign traffic ports and map them to the right activity emulation To do so dh Add 13 Select Ports menu from 255 14 Add the IP address or domain name of the allocated chassis and wait for IxLoad to refresh and display the available port resources 15 Select the options appropriate for your test configuration and network infrastructure map them to the corresponding NetTraffics Once complete save the file to disk for further use PN 915 2611 01 Rev H June 2014 229 Test Case Measuring Quality of Experience for Mult
290. ve Verify that the Ramp Up Value and Ramp Up Interval parameters available in the timeline configuration of the test objective creates users at a rate equal to or higher than the CPS rate configured by the objective June 2014 43 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Test Variables Test Tool Variables Table 12 Parameter Name Current Value Additional Options Calls Attempted Rate Up to 3 000 cps per XMV12 RTP card User Constraint 8 000 Up to 96000 per XMV12 RTP card Objective constraint for the Calls ear Concia Talk Time constraint call hold time Attempted per Second Scenario Call Hold constraint Network Configuration IP Only Static IPs IP DHCP IP IPsec IP PPPoE SIP Transport SIP over UDP SIP over TCP SIP over TLS TCP Codec parameters for the negotiated audio CODEC type Sa a Opt a G 729 G 723 G 726 iLBC packet size amp frequency y Mix with data protocols for Norme ided Any combination of data protocols example FTP HTTP Telnet supported by IxLoad IMS Call Flows PRACK Voice Call Flow ou OAOT Standard Custom Call Flows specific to a call flow particular DUT Secure RTP Off On IP Mapping rules 1to1 1toN Nto1andNtoN TOS DSCP marking for SIP amp RTP On individual settings for SIP amp RTP DUT Test Variables Table 13 Parameter Name Current Value Additional Options v4 Transport Protocol SIP UDP SIP TCP SIP TLS PN 915 2611 01 Rev
291. ver Figure 58 H323 Receive_Call Codec Settings page PN 915 2611 01 Rev H June 2014 63 Test Case Determining the Maximum Number of Concurrent Calls Configuring the RTP Settings for the Receive_Call Activity 1 Click the RTP tab 2 Select the Enable advanced stats calculation check box 3 Select the Per Stream Statistics check box 4 Select the Enable HW Acceleration check box to allow 8 000 RIP streams per port Traffice Receive_Call VoIPH323 Peer Dia Plan 323 Terminal Capit Hardware acceleration Audio video port 10000 65535 2 RTCP Calculate advanced statistics Per Stream Statistics verify all settings Restore defaults Figure 59 Configuration example for the RTP Settings page Configuring the Audio Settings for the Receive_Call Activity 1 Click the Audio tab 2 Verify that the Enable audio on this activity check box is selected 3 Set the duration of the Play for to 5 minutes 4 Select the Perform MOS check box to also automatically select the Calculate One Way Delay check box Traffice Receive _Call VoIPH323 Peer Enable audio on this activity Cif unchecked all audio script Functions will be SKIPPED Play Settings _ Enable jitter buffer Clip US_O042 way Buffer ciao Format PCM Duration 32785 ms Size 524556 bytes lse compensation Output level 20 dem Type OF Service Units E TOS DSCP A Value Perform MOS Calculate One Way Delay _ Generate silence
292. ver the network and affects the QoS As a best practice it is good to over commission the video bandwidth rate enforcements and allow up to 20 overhead if needed Another important aspect to be mentioned is the video tendency to send a large number of packets in short burst instead of an even distribution thus tolerances are accommodated on the entire network infrastructure Without these optimizations the effect will be a poor image quality received mostly because of the delay and jitter introduced by passing at a constant rate the peaks of data through the various devices queues As for the Audio media encoding telepresence mandates the use of MPEG 4 Low Delay Audio Coder AAC LD a powerful audio compression format that assists both the perceptual audio coding and the low delay necessary for bidirectional audio communication The codec enforces a maximum logarithmic delay of 20ms and has a good support for speech and music encoding There are several bitrates that the encoder can use starting from 32Kbits but compared with other speech codecs the quality improves with the increase of bitrate The challenges of Preserving the QoS for Telepresence For most of the telepresence deployments the network pass through is the most significant source of delay and jitter This is a fixed characteristic since the time that the signal takes to propagate from one point to the other can increase dramatically when there are bottlenecks of bandwidth in betw
293. verse translation to the IP header In this way the destination IP address and port are updated with the IP address and port of the IP phone SRCIP SRC PORT DESTIP DEST PORT 193 16 148 244 192 168 1 100 200 OK SIP 2 0 Via SIP 2 0 UDP 193 16 148 244 5060 From lt s1p 8184443118 myserviceprovider com gt tag 3261c4561 To Alice lt sip 8184443118 myserviceprovider com gt Contact Alice lt si1p 8184443118 192 168 1 100 5060 gt Content Length 0 Expires 180 Figure 136 Structure of message 6 200 OK The IP Phone receives the 200 OK response and the registration is completed SBC s solution for permitting external calls To allow incoming calls the SBC must maintain an open pinhole through the CE NAT gateway This can be achieved using different methods such as forcing the IP phone to re register at shorter intervals for example 30 seconds or by sending a periodic OPTIONS requests from the SBC to IP phone The pinhole will be kept open regardless of the response type from the IP Phone 200 OK versus 4xx response A better approach would use TCP as the transport protocol All external calls are received from the SBC Hence even if the edge of the service provider has a firewall installed the security policy can be updated to allow any SIP message received from the IP address of the firewall which is well known because the service provider shares this information with their customers PN 915 2611 01 Rev H J
294. vide the same communication services without installing a local IP PBX in each remote location Another benefit comes from simplified scalability with IP telephony businesses can control and avoid underutilized equipment New users may be added to existing locations one at a time rather than buying or leasing equipment that may remain underutilized VoIP Challenges To offer a good alternative to PSTN services VoIP must provide the same speech quality and service reliability as customers are accustomed to receiving from legacy circuit switched networks PSTN is well known for its five nines in reliability 99 999 percent uptime which corresponds with 5 minutes of downtime a year The quality of voice in PSTN networks is PN 915 2611 01 Rev H June 2014 1 Voice over IP referred to as toll quality this corresponds to a mean opinion score MOS score of 4 00 measured on a scale from 1 fair to 5 excellent as specified by the ITU P 800 standard Network Requirements for Toll Quality The network itself plays a critical role in delivering toll quality voice IP telephony requires the network to be designed to deliver enough bandwidth and to meet specific latency jitter and packet loss requirements While data traffic is less impacted in networks with high delay jitter and packet loss voice communication is significantly impacted because it requires an end to end communication and has real time constraints Bandwidth Requirem
295. view 5 Has the QoS for media met the expected quality Check the RTP MOS RTP QoS RTP Advanced QoS RTP Jitter Distribution RTP Consecutive Lost Datagram Distribution and RTP Streams statistic views Table 6 MOS statistics Statistic Name ETEO Questions How do the last values reported by the RTP MOS Best and RTP MOS Worst RTP MOS Best compare with each other RTP MOS Worst How does the RTP MOS Worst score compare with the max theoretical score for the CODEC used Are any times without an instantaneous RTP MOS Instant i MOS value Best Avg Worst How frequent are the changes in the instantaneous MOS values How do the MOS per Call statistics compare with the RTP MOS Best and RTP MOS Worst statistics RTP MOS Per Call Best Avg Worst PN 915 2611 01 Rev H June 2014 41 Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems Table 7 Basic RTP QoS statistics see RTP QoS and RTP Advanced QoS statistics views Statistic Name VEE O Questions 1 Are there any differences between RTP Packets Sent and RTP Packets Received 2 Does the difference match the value of RTP Lost Packets RTP One Way Delay us ee 1 Is the One Way Delay higher than 100 ms RTP Packets Sent RTP Packets Received RTP Packets Lost RTP Delay Variation Jitter us RTP Interarrival Jitter us 1 What is the max Delay Variation Jitter 2 What is the max Interarrival Jitter Table 8 RTP J
296. w definition and global controls of the emulated channels As this scenario is focused on the signaling part of the TDoS the following steps will detail the actions required to send forged SIP messages This exercise will assist in constructing several REGISTER messages with various headers altered Similar steps can be followed to construct custom SIP methods such as INVITE OPTIONS PING and others In addition to custom messages the channel call flow can be manipulated to allow a higher flexibility in terms of SIP endpoint state machine emulation Networks and Traffic Scenario1 Originate HH B Add Channel I Note Zo Workspace x Flow Scenario Channel 0 b Start E Stop Variable Set f Variable Test Sleep Figure 225 Accessing the Scenario Editor configuration tab PN 915 2611 01 Rev H June 2014 240 Test Case Telephony Denial of Service 4 From the left side menu of the Workspace navigate to the Procedure Library and expand the SIP option This will display all the predefined functions that come with the default installation of IxLoad They allow a seamless configuration of the endpoint state machine with predefined actions Navigate through the options up to the SIP MakeRegistration Authentication procedure Click on it and add the new selection on the right side screen in the Scenario Editor screen for Scenario Channel 0 Once the procedure has been added as shown below link it from the St
297. w with a Packet time of 20 ms 160 bytes per frame PN 915 2611 01 Rev H June 2014 59 Test Case Determining the Maximum Number of Concurrent Calls Configuring the RTP Settings for the Make_Call Activity 1 Click the RTP tab 2 Select the Enable advanced stats calculation check box 3 Select the Per Stream Statistics check box 4 Select the Enable HW Acceleration check box to allow 8 000 RTP streams per port Traffic Make Call VoIPH323 Peer Dial Plan J H523 Terminal Capabity Hardware acceleration Audio Video port 10000 65535 2 _ ETEF Calculate advanced statistics Per Stream Statistics verify all settings Restore defaults Figure 53 Configuration example for the RTP Settings page Configuring the Audio Settings for the Make_Call Activity 1 Click the Audio tab 2 Verify that the Enable audio on this activity check box is selected 3 Set the duration of the Play for to 5 minutes 4 Select the Perform MOS check box to also automatically select the Calculate One Way Delay check box Trafficl Make Call VoIPH323 Peer Enable audio on this activity Cif unchecked all audio script Functions will be SKIPPED Play Settings _ Enable jitter buffer Clip US_ 042 way Format PCM Duration 32785 ms Size 524556 bytes Output level 20 dBm C Type OF Service Units of Channels TOSIDSCP Perform MOS Calculate One Way Delay _ Generate silence Figure 54 H323 Make_Call Configura
298. while Bob will continue to speak until the interruption is heard For Alice the effect of the delay is perceived as ignorance from Bob s side Hence she may stop talking On the other side Bob finally hears Alice s interruption and stops as well Both Bob and Alice remain silent and then both may start talking or stop talking To meet the toll quality requirement the one way delay must be kept below 100 ms A one way delay value of 150 ms makes its presence noticeable but they can still have an acceptable conversation In networks where VoIP interacts with PSTN the VoIP user may hear an echo if the PSTN network is not correctly tuned The round trip delay directly affects the effect of echo Hence the higher the roundtrip value the more annoying the effect of the echo ITU G 114 provides recommendations for network latency PN 915 2611 01 Rev H June 2014 2 Voice over IP Jitter Jitter represents the variation of delay as measured by the receiver Users perceive jitter as speech degradation Many devices implement jitter buffers that can remove the jitter effect A jitter buffer temporarily stores arriving packets to minimize delay variations and discards the packets that arrive too late If a jitter buffer is too small then an excessive number of packets may be discarded which can lead to call quality degradation If the jitter buffer is too large then an additional delay is added to the conversation which can lead to difficulties i
299. work settings accordingly to the particular test topology This will be the IP address of the emulated SIP server The same value should be set as the SIP Proxy IP address in the Caller and Callee activities The IP addresses of the other ranges under SIP_Proxy NetTraffic may remain unchanged while these are internal IP addresses used just to route the messages between the SIP Cloud to the SIP activities The number of IP addresses for the range IP R3 should be equal with the number of ports number of Ixia pots used to emulate the SIP Proxy 1 in this case The number of IP addresses of other ranges should be equal with the number of channels defined in Test Objective 1000 in this example PN 915 2611 01 Rev H June 2014 175 Test Case Determining the Performance of a Session Border Controller Originate aS Terminate pius SIP Prox Gar Traffic SIP_Caller m j Caller ae paa SIF_Frosy 2 S Filter 3 TCP 3 Settings 3 GratARP 3 DNS 3 TEER A MAC VLAN 3 Ethernet 3 r i Unconfigured 40 40 50 1 16 0 0 0 1 Increment every IP R4 Unconfigured IPw4 40 40 100 1 16 0 0 0 1 1000 0 0 0 0 0 0 0 0 Increment every IP RS Unconfigured IPy4 40 40 110 1 i6 0 0 0 1 1000 0 0 0 0 0 0 0 0 Increment every IP R6 Unconfigured IP 40 40 120 1 16 0 0 0 1 1000 0 0 0 0 0 0 0 0 Increment every 5 el TF R 7 Inranfin red TPa4 4 40 150 1 1A FI 1000 ALTA A Tnrrement eer Figure 165 Set the I
300. xample 1 hour and the available number of channels phones cannot achieve the desired rate However when the calls are generated they should be attempted at the constant call rate Requirement Example using a pool of 8 000 phones and calls with a talk time of 1 hour generate a steady call rate of 100 calls per second In this case the steady rate will be achieved only once every hour for a duration of 80 seconds 8 000 phones 100 cps 80 sec If the requirement imposes 100 cps with 1 hour talk time then at least 3 600 sec x 100 cps 360 000 phones will be required Calls Attempted per Second ge oO Q w w S w Qa co wv a E Vv lt 2 T Q 30 Time hrs Figure 20 Example of constant call rate with limited number of channels available and long Talk Time The duration of media functions can be specified at the activity level or at the function level in the test scenario We recommend you to use the activity level settings because these are exposed to TCL API in automation environment the control of call duration will be easily done without opening the test Scenario Editor The following steps show how to set the parameters of Voice Session functions in the Scenario Editor to use the setting at the activity level 1 In Network1 select the SIP Peer activity labeled Make_Call 2 The configuration page displays Scenario Editor 3 Click Full Screen in the Scenario Editor toolbar E a m
301. ximum number of concurrent calls H 323 example This example assumes the configuration instructions starting with the SIP configuration because the test objective will require modifications as well This example starts from the configuration IxLoad Voice SIP Call Setup Rate rxf resulted at the end of the Test Case Determining the Max Call Setup Rate for SIP Based Devices and Systems PN 915 2611 01 Rev H June 2014 146 Test Case Using VoIP to Measure NAT PAT Performance Updating the Test Scenario to Generate Voice and Video Traffic 1 2 Open the configuration IxLoad Voice SIP Call Setup Rate rxf Click the MakeCall activity and then click Save As to save the test scenario under the name SIP_NAT_MakeCall ReceiveCall EndCall with RTP tst Save the configuration main menu File Save As under the name IxLoad Voice SIP NAT rxf Select either one of the two activities Make_Call or Receive_Call ser Xe Orignate gt DUT t Terminate Fl sewn Oe oT Make Call go ae iy gpr JE Traffic Ee j Receive_Call Traffici Make Call YoIPSip Feer Scenario Execution piat Plan StP Automatic Ts Cloud Codecs RTP Audio video Fax 7 98 Fax 7 90 SRTP other Coe gt Mon SOB X o o A Ehe r eee S Figure 141 Test Configuration selection of Scenario Editor The configuration page displays Test Scenario Editor Click Full Screen available i
302. yed in organizations or public sector services The reason is not always obvious but the effects on the service are severe and usually have a long recovery time This form of malicious traffic has several levels of impact on the recipient of the attack including damage to the public image and the immediate financial losses due to increased customer churn Also there are the added costs for analysis of the security breach and from the prevention systems deployed after the incident Usually these are over architected to prevent future worst case scenarios and the costs penalties are significant Another long term cost is the added investment in recovering the company image and replacing lost customers In most of the cases the digital telephony has a low level of security and as soon as network convergence is obtained for optimum user experience and system capacity no additional changes are made to avoid disruption of the functional system Additional levels of voice security enforcements have immediate effect on the capacity or quality and if the effect is significant those changes are reverted and rarely re evaluated and enforced This paradigm of not constantly improving the security of a system that works in production leaves open doors for malicious traffic and may compromise the entire system stability Recently the Department of Homeland Security announced that investigations are active for hundreds of service attacks and a third of the

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