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Avaya IP Telephony Configuration Guide

1. Field Value Description H323AliasName lt unique name gt This is the unique name that identifies your Business Communications Manager as an H 323 endpoint CDP Domain Name lt choose name from list gt If your system is using a CDP dialing plan choose the CDP domain name for the Business Communications Manager Tandem Endpoint lt choose name from list gt This is the name of another H 323 endpoint Picking a name in this field provides a tandem endpoint 5 Click Create H323 Setting the H 323 Endpoint Dialing Plan All dialing plan information must be identical on all H 323 endpoints using the gatekeeper Follow these steps to set the dialing plan into the Gatekeeper Admin tool A OO N a Select GK Standby DB Admin Select NumberPlanEntries Select Create Ensure that the Endpoint you select is the one for which you want to create a numbering plan entry Click Select P0993474 03 Chapter 5 Configuring VoIP trunks 111 6 Ensure that the following fields are set Table 25 CSE 1000 H 323 dialing plans Field Value Description Number digits This is the unique number that identifies the Business Communications Manager Type choose from list This is the TON Type of Number or NPI Numbering Plan Identifier for the endpoint EntryCost digits 1 255 gt This value determines which destination the gatekeeper will deliver to if the leading digits are the same fo
2. Table 9 IP terminals general record fields Continued Field Value Description Jitter Buffer None Choose one of these settings to change the default jitter buffer size Auto e NONE Minimal latency best for short haul networks with Small good bandwidth Medium AUTO Business Communications Manager will dynamically Large adjust the size e SMALL Business Communications Manager will adjust the buffer size depending on CODEC type and number of frames per packet to introduce a 60 millisecond delay e MEDIUM 120 millisecond delay LARGE 180 millisecond delay For information about choosing a Jitter Buffer refer to Choosing a Jitter Buffer on page 43 4 Goto Installing i series telephones on page 43 Choosing a codec The default codec is used when an IP client has not been configured to use a preferred Codec Refer to the next section for individual IP client Codec settings If the default Codec is set to AUTO the Business Communications Manager will choose the appropriate CODEC when an IP client makes a call For example if both endpoints of the call are IP telephones on the same subnet the Business Communications Manager chooses G 711 for maximum voice quality If the telephones are on different subnets the Business Communications Manager will choose G 729 to minimize network bandwidth consumption by voice data packets Note If the IP telephones are using VoIP trunks for the call the codec set
3. IP Telephony Configuration Guide 56 Chapter3 Installing IP telephones 3 Select the feature you want to modify and right click or click on the Configuration menu item then select the action you want to perform Performance Fault Modify parameters Deregister DN Force firmware download Reset Hot Desking Password Modify Feature Delete Feature The Telephony Features list screen appears Figure 10 Add Modify Telephony Features List Telephony Features List Feature Index Ft Feature Name SWCA 1 Feature Code Format 0123456789 4 Enter or change the Feature Name and corresponding Feature Code in the appropriate fields 5 Click Save The features list appears Notice that the system assigns a Feature Index number adding the feature to the bottom of the list Using the Services button to access features The IP telephone has a limited number of memory buttons that can be configured with lines or features however a soft features menu also can be accessed by pressing the Services button Ce e Use the arrow buttons or the Page and Page display keys to move quickly through the list e Use the up and down directional buttons to scroll to each feature P0993474 03 Chapter 3 Installing IP telephones 57 e Press the Select display key to activate the feature then use the feature as you normally would For example if you selected Call Forward enter the number you to which you wa
4. P0993474 03 Efficient Networking 123 Table 28 VoIP Transmission Characteristics for unidirectional continuous media stream Continued Codec Type Payload Size IP Packet Ethernet B W PPP B W FR B W ms Bytes Bytes kbit s kbit s kbit s G 729 10 10 50 60 8 41 6 47 2 8 kb s 20 20 60 34 4 24 8 27 6 30 30 70 25 6 19 2 21 1 G 723 1 30 24 64 24 0 17 6 19 5 6 3 kb s G 723 1 5 3 kb s 30 20 60 22 9 16 5 18 4 Notes 1 Gray background indicates payload sizes used by Business Communications Manager 3 0 for transmission Other values listed indicate payload sizes that the Business Communications Manager 3 0 can receive 2 Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter frame gap The peak bandwidth and average bandwidth requirements for a normal two way call must take into account the affects of full and half duplex links and the affects of silence suppression Refer to the tables in the next two sections below and to Table 30 on page 125 for voice Gateway bandwidth requirements Peak bandwidth is the amount of bandwidth that the link must provide for each call Considering voice traffic only the number of calls a link can support is Number of Calls Usable Link Bandwidth peak Bandwidth per call The average bandwidth takes into account the affects of silence suppression which over time tends to reduce bandwidth requirements to 50
5. Chapter 5 Configuring VoIP trunks 107 Table 19 Local Gateway IP interface fields Continued Field Value Description Example In the following example the Business Communications Manager is assigned an E 164 and an H323 Identifier Alias Names TEL 76 NAME bcm10 nortel com In the following example the Business Communications Manager is assigned a public dialed number prefix of 76 a private DCP number of 45 and an H323 Identifier alias Alias Names PUB 76 CDP 45 NAME bcm10 nortel com Note E164 or NPI TON alias types are commonly used since they fit into dialing plans A Business Communications Manager alias list should not mix these types Also the type of alias used should be consistent with the dialing plan configuration Use the same alias on all Business Communications Managers within a networked system Registration TTL Default 60 seconds This TimeToLive parameter specifies the intervals when the VoIP gateway sends KeepAlive signals to the gatekeeper The gatekeeper can override this timer and send its own TimeToLive period Gateway Protocol None If you are using an MCDN protocol on the IP trunk select SL1 SL1 Note You require a keycode for this protocol GWProtocol Otherwise use None Gateway Protocol G wProtocol hd none SL1 BWProtocol These fields are mandatory when you use Radvision ECS 2 1 0 1 These fields are optional when you use Radvision ECS
6. l l 150 Appendix E Oi isbe nl saci coi nt Sees ei D r 153 rupes M ETT T P EM 153 Measu nng INUANSI QOS 2 004 Geto s PPESZCEESLEAQEEGEESTIqIIRESNIXOSTA SERI ERAT quA 155 Measuring end to end network delay selseeee eh 155 Measuring end to end packetloss 0c eee tee elles 156 Recording OUS access quere BREE CONNER ee ER ew PERE SENSE ERE SEN eR REM 156 Adjusting Ping measurements sici weeds edie dad dea ded yous es yas 157 Adjustment TON processing iaeic ined a uaxs iier biessXPedGdescduqaddd 157 LEG PACKETS MeT 157 Measurement PROGR iucauck Leopgokuba sh Gp a ho BE ECRIRE RC EORUCEUE UA RE RE e REMO eden O E 158 Other measurement considerations 25 ce0e25 8 sees es Rh ERR ARR AES 158 Decision does the intranet meet IP telephony QoS needs 159 implementing QoS in IP networks cli esc cue duke Es rie ORE eon ciudad Hg A 159 Hr ud ccc een nad ae ton Beane EREE rE RRE EREEREER eae aR Re ton md ce 160 TOP TRIE DOVII sic45 der REY dade gente OE Rhee Tage CROP AS Ro AP eS 160 Business Communications Manager router QoS support 20 eee ee 161 Network Quality OT BOIVIGB cuusussceek heck esee Oe wed coe ow oa eR OR EROR GR Rh GR 161 Nelwork monet 2 iccuo reste datae bee ee eh uq xU Edible aol bd 162 Quality of Service parameters as sce cece eee reac ERRARE sea Ree Eee RS 162 PACKS IGG ay ces8eas Eier RE C 28h hs E E bee
7. Configuring PSTN fallback on page 80 7 Click on the line pool that you selected as the VoIP line pool The Pool screen appears as shown in the figure below Figure 19 Line pool access code setting Access code 7 8 Enter a unique access code for this line pool IP Telephony Configuration Guide 78 Chapter5 Configuring VoIP trunks Ensure that no other line pools use this access code Also ensure that this access code is not used for any other type of code such as destination codes or DISA DNs Configuring telephones to access the VoIP lines For each telephone that will be allowed to use the VoIP lines you must add that line pool to the DN record 1 In Unified Manager open Services Telephony Services System DNs Active Set DNs DN XXX Line Access DN XXX is any DN that you want to allow to use VoIP trunking 2 Click Line Pool Access 3 Click Add The Add Line Pool Access dialog appears 4 Typethe letter of the VoIP line pool 5 Click Save 6 Repeat this procedure for every telephone you want to allow to use VoIP trunks Configuring a remote gateway This section explains how to configure the Business Communications Manager to communicate with other Business Communications Managers and or other VoIP gateways such as Meridian ITG The remote gateway list must contain an entry for every remote system to which you want to make VoIP calls Note Gatekeeper If your system is controlled by a
8. Nortel Networks i2050 Software Phone 1 0 x Nortel Networks M1 ITG ITG2 X 26 Microsoft NetMeeting 3 0 Symbol NetVision Telephone 03 50 12 01 00 24 or greater Business Communications Manager IP Telephony interoperates with the Gatekeeper applications Radvision ECS 2 1 0 1 and CSE 1000 which conform to the specifications in the following tables Table 33 Engineering specifications Capacity 1 to 8 ports Voice compression G 729 CS ACELP 8 kbit s G 711 PCM 64 kbit s u A law G 723 1 MP MLQ 6 3 kbit s or ACELP 5 3 kbit s supports plain Annex A and Annex B Silence compression G 723 1 Annex A G 729 Annex B IP Telephony Configuration Guide 144 Interoperability Table 33 Engineering specifications Capacity 1 to 8 ports Echo cancellation 48 ms tail delay In band signaling DTMF TIA 464B Call progress Speech path setup methods Call Initiator e H 323 fastStart Call Terminator e H 323 slowStart e H 323v2 fastStart End to end DTMF signaling digits 0 9 and fixed duration tones only Table 34 Supported voice payload sizes Codec Receive transmit to M1 ITG Receive transmit to others G 711 Highest supported by both ends up to 20 ms 30 ms in 10 ms increments G 723 1 30 ms 30 ms G 729 Highest supported by both ends up to 20 ms 30 ms in 10 ms increments Speech path setup methods Business Communic
9. e The gateways interface with the intranet the service provided by the intranet is best effort delivery of IP packets not guaranteed QoS for real time voice transport IP telephony translates the QoS objectives set by the end users into IP adjusted QoS objectives The guidelines call these objectives the intranet QoS objectives IP Telephony Configuration Guide 154 Quality of Service Figure 52 Relationship between users and services Delay variation Business Communications Manager IP telephony parameters Fallback threshold Silence compression Codec Echo cancellation Non linear programming I I Business Communications I l Manager VolP Corporate intranet Deliver voice fax service Deliver IP service User oriented QoS Network QoS metrics Roundtrip conversation delay One way delay Clipping and dropout Packet loss echo The IP gateway can monitor the QoS of the Intranet In this mode two parameters the receive fallback threshold and the transmit fallback threshold control the minimum QoS level of the intranet Fallback thresholds are set on pair per site basis The QoS level is aligned for user QoS metrics to provide an acceptable Mean Opinion Score MOS level The administrator can adjust the fallback thresholds to provide acceptable service to the users The settings in the following table indicate the quality of voice service IP telephony periodi
10. Checking IP server status on page 52 Preparing your system for IP telephone registration When you install an IP telephone on a Business Communications Manager you must activate terminal registration on the Business Communications Manager If this is your first installation you need to set the general parameters for IP registration Note For the simplest installation possible set telephone Registration and Auto Assign DNs to ON and leave Password blank IP telephones installed on the system LAN will connect and boot up without manual registration Warning Nortel cautions that leaving your system in this state may pose a security risk P0993474 03 Chapter 3 Installing IP telephones 41 1 InUnified Manager open Services IP Telephony and Nortel IP Terminals 2 Select the General tab The General screen appears as shown below Figure 5 Set registration properties E dit Performance Fault Report Tools Logoff 47 565 138 52 em ources ices Telephony Services P Telephony System Configu H 323 Terminal H 323 Trunks PortRanges Sall Detail Recordin AN CTE Confiaura General 3 Summary General Registration OFI Password Auto Assign DNs Advertisement Logo Default Codec Default Jitter Buffer Nortel Networks AUTO Z AUTO w 3 Use the information in the table below to set up your IP terminals general information
11. Reset Hot Desking Password Add Feature Modify Feature Delete Feature 5 Select Force Firmware Download A dialog appears asking if you want to confirm that you want to proceed 6 Click the Yes button The firmware download begins The system drops any active call on that telephone and downloads a new firmware load into the selected telephones The telephones will be unusable until the download is complete and the telephones have reset Note In order not to saturate the IP network with download packets the system will only download up to five IP telephones at any given time Telephones requiring download will show a Unified Manager status of Download Pending and the UNISTIM Terminal Proxy Server UTPS will initiate download as resources become available IP Telephony Configuration Guide 62 Chapter3 Installing IP telephones Deregistering DNs for IP telephones You can deregister selected telephones from the Business Communications Manager and force the telephone to go through the registration process again j Warning Once this feature is activated all active calls are dropped To deregister a DN for a telephone 1 Figure 13 Configuration Performance Fault Report Tools Logoff In the Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary appears Click on the IP Terminal Status tab Select the IP Terminal with the DN you want to dereg
12. ip address The set IP must be a valid and unused IP address on the network that the telephone is connected to NETMASK subnet mask address This is the subnet mask This setting is critical for locating the system you want to connect to DEF GW ip address Default Gateway on the network i e the nearest router to the telephone The router for IP address W X Y Z is usually at W X Y 1 If there are no routers between the telephone and the Business Communications Manager network adaptor to which it is connected for example a direct HUB connection then enter the Published IP address of the Business Communications Manager as the DEF GW If the IP telephone is not connected directly to the Published IP address network adaptor set the DEF GW to the IP address of the network adaptor the telephone is connected to For information on setting the published IP address of the Business Communications Manager see Defining published IP address on page 35 1 IP ip address This is the Published IP address of the first Business Communications Manager that you want to register the telephone to S1 PORT Default 7000 This is the port the telephone will use to access this Business Communications Manager S1 ACTION Default 1 1 RETRY COUNT digits between 0 and 255 Set this to the number of times you want the telephone to retry the connection to the Business Communications Man
13. Table 9 IP terminals general record fields Field Value Description Registration On Set this value to ON to allow new IP clients to register with the Off system Password 10 alphanumeric gt This is the password the installer will enter on the IP telephone to Default bcmi connect to the Business Communications Manager If this field is left blank no password prompt occurs during registration Auto Assign DN On If set to ON the system assigns a free DN as a set requests Off registration It does not prompt the installer to enter a set DN Note Registration must be ON and Password must be blank If set to OFF the installer receives a prompt to enter the assigned DN during the programming session Advertisement Logo alphanumeric string Any information in this field appears on the display of all IP telephones For example your company name or slogan Default Codec Auto G 711 aLaw G 711 uLaw G 729 G723 G 729 VAD G 723 VAD If the IP telephone has not been configured with a preferred codec choose a specific codec that the IP telephone will use when it connects to the system If you choose Auto the IP telephone selects the codec For information about choosing a codec refer to Choosing a codec on page 42 If you are unsure about applying a specific codec ask your network administrator for guidance IP Telephony Configuration Guide 42 Chapter3 Installing IP telephones
14. e If you are prompted for a DN enter the DN you want assigned to this telephone and press OK When the telephone registers it downloads the information from the Business Communications Manager IP Telephony record to the telephone configuration record This might include a new firmware download which occurs automatically If new firmware downloads the telephone display indicates the event Note If the telephone displays a prompt that indicates it cannot find the server follow the instructions in Configuring telephone settings to enter the specific network path Troubleshooting an IP telephone on page 48 describes other possible prompt messages Once registration has completed you do not need to go through the registration steps described above unless you deregister the terminal For information about setting the registration settings see Preparing your system for IP telephone registration on page 40 Configuring telephone settings If you are not automatically registered to the Business Communications Manager you can configure your telephone settings to allow you to access a system on the network You will also need to perform these steps if your IP telephone is not connected to the same LAN to which the Business Communications Manager is connected Follow these steps to access the local configuration menu on an 12002 or an 12004 telephone 1 Restart the telephone by disconnecting the power then reconnecting the power Af
15. 1 The installer sets up 3321 as the Control set for each VoIP line so that the VoIP route can be manually activated The installer sets the published IP address In this case the public data network PDN is connected to the LAN 2 connection therefore the installer sets the published IP address to LAN 2 This is the address that devices on the PDN will use to locate the system The installer configures the media for the system using the following settings e The first preferred codec is set to G 729 e Silence Compression is turned on e Jitter Buffer is set to medium The installer puts the first eight VoIP lines into line pool O Any line pool can be used as long as all of the lines in the pool are VoIP The installer does not set an access code for the line pool because the access code would not work with fallback Instead the line pool will be accessed using destination digits after the installer sets up PSTN fallback For each set on the system DNs 3321 to 3331 the installer gives the set access to line pool O The installer sets up a remote gateway for the Santa Clara Business Communications Manager using the following settings e Destination IP 47 62 54 1 This is the published IP address of the Ottawa Business Communications Manager e QoS Monitor Enabled This must be enabled for PSTN fallback to function e Transmit Threshold 3 0 This is a MOS value that ensures that the VoIP lines are used as long as th
16. Device that converts serial data from a transmitting terminal to an analog device for transmission over a telephone channel Another modem converts the signal to serial digital Noise P0993474 03 Glossary 169 network diagram This is a physical drawing description of how the local network works to which your Business Communications Manager will be connected It also includes information about the Business Communications Manager requirements such as public and or private IP addressing DHCP requirements and quality of service availabilities Where possible it should include information about the public networks and any changes or adjustments required by the network or the Business Communications Manager for compatibility Nortel NetVision Phone Administrator NVPA This is the Business Communications Manager specific application that is used to configure features and system information into the NetVision handsets This application is included on the Business Communications Manager database The latest application can be obtained at http www symbol com services downloads nvfirmware2 html The serial cable required to update the programming of the handset can be purchased from Purchased from Symbol at lt http symbol com gt part number 25 20528 01 packet Group of bits transmitted as a complete package on a packet switched network packet switched network PSTN A telecommunications network based on packet switching tech
17. Ping e Traceroute e Sniffer Ping Ping Packet InterNet Groper sends an ICMP Internet Control Message Protocol echo request message to a host It also expects an ICMP echo reply which allows for the measurement of a round trip time to a selected host By sending repeated ICMP echo request messages percent packet loss for a route can be measured Traceroute Traceroute uses the IP TTL time to live field to determine router hops to a specific IP address A router must not forward an IP packet with a TTL field of 0 or 1 Instead a router discards the packet and returns to the originating IP address an ICMP time exceeded message Traceroute sends an IP datagram with a TTL of 1 to the selected destination host The first router to handle the datagram sends back a time exceeded message This message identifies the first router on the route Then Traceroute transmits a datagram with a TTL of 2 Following the second router on the route returns a time exceeded message until all hops are identified The Traceroute IP datagram has a UDP Port number not likely to be in use at the destination normally gt 30 000 The destination returns a port unreachable ICMP packet The destination host is identified Traceroute is used to measure round trip times to all hops along a route identifying bottlenecks in the network IP Telephony Configuration Guide 142 Network performance utilities Sniffer Sniffer is not provided with the Busi
18. both Business Communications Managers must list each other as a Remote Gateway and QoS Monitor must be enabled on both systems Updating the QoS monitor data To update the table with the most current values From the View menu select Refresh P0993474 03 Chapter 5 Configuring VoIP trunks 101 Viewing QoS monitoring logging QoS monitor can be configured to log data The process for setting up logging is described in detail in the Programming Operations Guide The following steps explain how to view the log On the Unified Manager navigation tree click the keys beside Services and Qos Monitor Click the Mean Opinion Score heading Click the Logging tab The Logging screen appears 4 Onthe Tools menu click Display Log The Mean Opinion Score Log File screen appears 5 Close the browser window when you are finished viewing the log file Port settings In some installations you may need to adjust the port settings before the Business Communications Manager can work with other devices This section includes information about e Using firewalls on page 101 e Port settings for legacy networks on page 103 Using firewalls Firewalls can interfere with communications between the Business Communications Manager and another device The port settings must be properly configured for VoIP communications to function properly Using the instructions provided with your firewall ensure that communications using th
19. is used to separate fields ASCII is used to separate Primary from Secondary Business Communications Manager information ASCII is used to signal end of structure iii jjj kkk 111 ppppp identifies IP port for server ASCII encoded decimal aaa identifies Action for server ASCII encoded decimal range 0 255 rrr identifies retry count for Business Communications manager ASCII encoded decimal range 0 255 This string may be NULL terminated although the NULL is not required for parsing Notes aaa and rrr are ASCII encoded decimal numbers with a range of 0 255 They identify the Action Code and Retry Count respectively for the associated Business Communications Manager Internal to 12004 they will be stored as 1 octet 0x00 0xFF Note that these fields must be no more than three digits long the Business Communications Manager is always considered the Primary server the second server always considered Secondary if only one Business Communications Manager is required terminate primary TPS sequence immediately with instead of e g Nortel i2004 A iii jjj kkk lll ppppp aaa rrr valid options are one Business Communications Manager or two Business Communications Managers 0 3 not allowed IP Telephony Configuration Guide 52 Chapter3_ Installing IP telephones e Action code values 0 reserved 1 UNIStim Hello currently only this type is a valid choice 2 254 reserved 255 reserved e iii jjj kkk
20. of the continuous transmission rate The affects of silence suppression on peak bandwidth requirements differ depending on whether the link is half duplex or full duplex See Appendix B Silence compression on page 135 for more information When engineering total bandwidth requirements for LANs and WANs additional bandwidth must be allocated for data Refer to standard Ethernet engineering tables for passive 10BaseT repeater hubs Refer to the manufacturer s specification for intelligent 10BaseT layer switches WAN links must take into account parameters such as normal link utilization and committed information rates IP Telephony Configuration Guide 124 Efficient Networking Bandwidth requirements on half duplex links The following table provides bandwidth requirements for normal two way voice calls on a half duplex link for a variety of link protocols codec types and payload sizes Table 29 Bandwidth Requirements per Gateway port for half duplex links Ethernet B W PPP B W FR B W Codec Payload Silence Silence Silence Type Size NoSP Suppression No SP Suppression No SP Suppression ms peak peak Avg peak peak Avg peak peak Avg kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s G 711 10 233 6 233 6 233 6 195 2 195 23 195 23 206 4 206 43 206 43 64 kb s 20 180 8 180 83 180 83 161 6 161 63 161 63 167 2 1
21. on page 121 e Network engineering on page 122 e Additional feature configuration on page 127 e Further network analysis on page 131 e Post installation network measurements on page 134 Determining the bandwidth requirements The IP network design process starts with the an IP telephony bandwidth forecast The bandwidth forecast determines the following e LAN requirements LAN must have enough capacity for the number of calls plus the overhead e WAN requirements WAN must have enough capacity for the number of calls plus the overhead Determining WAN link resources For most installations IP telephony traffic travels over WAN links within the intranet WAN links are the highest recurring expenses in the network and they are often the source of capacity problems in the network WAN links require time to receive financial approval provision and upgrade especially inter LATA Local Access and Transport Area and international links For these reasons it is important to determine the state of WAN links in the intranet before installing IP telephony Link utilization This procedure explains how to determine and adjust link utilization 1 Get a current topology map and link utilization report of the intranet A visual inspection of the topology can indicate the WAN links anticipated to deliver IP telephony traffic 2 Record the current utilization of the links that will be handling IP telephony traffic For example the
22. you leverage the untapped capabilities of your data infrastructure to maximize the return on your current network investment e Portability and flexibility Employees can be more productive because they are no longer confined by geographic location IP telephones work anywhere on the network even over a remote connection With Nortel Networks wireless e mobility solutions your phone laptop or scanner can work anywhere on the network where a an 802 11b access point is installed Network deployments and reconfigurations are simplified and service can be extended to remote sites and home offices over cost effective IP links Simplicity and consistency A common approach to service deployment allows further cost savings from the use of common management tools resource directories flow through provisioning and a consistent approach to network security As well customers can centrally manage a host of multimedia services and business building applications via a Web based browser The ability to network existing PBXs using IP can bring new benefits to your business For example the ability to consolidate voice mail onto a single system or to fewer systems makes it easier for voice mail users to network e Compatibility Internet telephony is supported over a wide variety of transport technologies A user can gain access to just about any business system through an analog line Digital Subscriber Line DSL a LAN frame relay asynchronous tra
23. 2 1 0 1 Notes about NPI TON aliases NPI TON aliases store dialed number prefixes as well as information about the type of number A dialed number c an be qualified according to its TON type of Number as well as its NPI numbering plan identification Nortel Networks recommends this format over the E 164 format for encoding dialed numbers and aliases registered with a gatekeeper When using a gatekeeper and attempting to place an outgoing VoIP trunk call ensure that the route and dialing plan configuration matches the NPI TON aliases registered by the destination with the gatekeeper These requirements are summarized in the following table Table 20 Route and Dialing Plan configurations for NPI TON Route DN type Dialing Plan used by calling gateway Alias configured for calling gateway Public Public PUB lt dialedDigitsPrefix gt Private Private Type None PRI lt dialedDigitsPrefix gt Private Type CDP CDP lt dialedDigitsPrefix gt Private Type UDP UDP lt dialedDigitsPrefix gt IP Telephony Configuration Guide 108 Chapter5 Configuring VoIP trunks Using Radvision ECS 2 1 0 1 as the gatekeeper When you use Radvision ECS 2 1 0 1 as the gatekeeper with the Business Communications Manager specifically with the FP1 Maintenance Release use the configurations described in this section For detailed information about Radvision and how to open and use the applicatio
24. 421 to 10 10 10 19 and returns this IP in an AdmissionConfirm to the Business Communications Manager Ottawa Business Communications Manager Ottawa sends the call Setup message for DN 421 to the gateway at 10 10 10 19 and the call is established If call signaling is set to Gatekeeper Routed and no pre granted ARQ has been issued 1 Business Communications Manager Ottawa sends an AdmissionRequest to the gatekeeper for DN 421 The gatekeeper resolves DN 421 to 10 10 10 17 Business Communications Manager Ottawa sends the call Setup message for DN 421 to the gatekeeper 10 10 10 17 which forwards it to the gateway at 10 10 10 19 The call is established IP Telephony Configuration Guide 114 Chapter5 Configuring VoIP trunks P0993474 03 115 Chapter 6 Typical network applications using MCDN This section explains several common installation scenarios and provides examples about how to use VoIP trunks and IP telephony to enhance your network Information in this section includes e Setting up MCDN over VoIP with fallback on page 115 e Networking multiple Business Communications Managers on page 117 e Multi location chain with call center on page 119 Business Communications Manager to IP telephones on page 120 Setting up MCDN over VoIP with fallback The MCDN networking protocol between a Meridian 1 and one or more Business Communications Managers works the same way as it does over P
25. 76 outgoing calls 76 port ranges legacy systems 103 port settings 101 PSTN fallback 80 PSTN fallback schedule 82 Published IP address 35 QoS monitor status 100 remote access warning 97 remote gateway 78 routing 82 setting up target lines 89 signaling method 104 silence compression 75 target lines 89 trunk capacity 130 using a gatekeeper 103 using firewalls 101 W WAN Business Communications Manager function 34 link resources 121 network engineering 126 Published IP address 35 Warning symbol 13 wireless IP 67 workstation prerequisites 37 P0993474 03
26. Chapter 5 Configuring VoIP trunks 109 Gatekeeper support for interoperability 6 Create a service configuration for ITG a Select the Services tab b Click on the Add button C Inthe Prefix field enter the unique telephone number that identifies the Meridian ITG system in the Business Communications Manager dialing plan 7 Define the ITG as a predefined endpoint a Select the Endpoints tab b Click the Add predefined button The Predefined Endpoint Properties dialog displays C Ensure the following fields are set Table 23 Radvision Predefined Endpoints Properties settings Field Value Description Endpoint Type Gateway Force Online Status check box selected Registration IP ip address This is the IP address of the Meridian ITG system Aliases Add Name Phone Number Name The name of the ITG that will be displayed Phone Number The number assigned to the ITG Radvision uses this number to identify calls to be routed to this ITG Allowed Services Allowed Disallowed Ensure the ITG service is on the list and is Allowed 8 Close the application 9 Run system tests to ensure the gatekeeper is routing calls correctly Using CSE 1000 as a gatekeeper Both the Business Communications Manager and the CSE 1000 must be set to the parameters described in this section for the gatekeeper to work effectively The CSE 1000 GK Admin tool is obtained fr
27. Enter a unique password This is what the user must enter on the handset to connect to the system from the handset You must enter at least four digits This is a mandatory field IP Address read only This field populates when the system assigns an IP address to the handset Status read only This field populates when the system registers the handset 4 Click Save Note Shortly after the H 323 Terminals record is saved the system moves the DN you specified to the Active DNs list If you have not already done so configure the DN record for user requirements If you are not sure about how to configure DNs refer to the Business Communications Manager 3 0 Programming Operations Guide for details about the various settings within this record Programming note Ensure that you choose Model PWiIs on the General screen Testing the handset functions When the handset is registered check the handset feature menu and test the handset to ensure it is working as you expected Refer to the NetVision Telephone Feature User Card for directions about using Business Communications Manager call features on the NetVision handset Updating the H 323 terminals record If you need to change the password for a NetVision telephone update the H 323 terminals record Follow these steps to update the H 323 Terminals record 1 2 3 4 c1 In the Unified Manager click on the keys beside Services and IP Telephony Click on H 323 Terminal
28. Figure 16 H323 Terminal list dialog icio estes dares dan aie ede VR RT REOR Ra 70 Figure 17 Media Parameters sucssuxamxenedae eR eda sehen Kaede sven Koes XAR 74 Figure 12 Tr nkg Line Cate ose oss sme mee pex Ra x b Rex ex REGE Rome 77 Figure 19 Line pool access code setting 520 c0cce rece eee eee ee eee TT Figure 20 Remote gateway dalog uuuaeau soe aepo cue ROS RERCRUR k DERRETE TIERE ES 79 kigure 2 PSTN fallback Magras 34 cix EE aod ERR ed RC RR RR A 80 Figure 22 VolP Routing Service inosus cscocexhooc eee bese owe PRESSE d de E des SR RC 82 Figure 29 Add route dialog ise ceo mee mme knee me degere aS Rh 83 Figure 24 VoIP schedule amp oossossakrdeksdotikbxedEbpeveebo4e dd eR e XE RR 85 Figure 25 Normal schedule routing information 0 0 eee eee ees 86 Figure 26 QoS Monitor field on the Remote Gateway screen 000 0 eae 88 Figure 27 Threshold fields on the Remote Gateway screen 000 e eee 88 Figure 28 Fallback Matres TEMS cswsncdevases aerate tthis ta Gee Heda Rm a pe le 88 Figura 29 Example PSTN TIllBSCK coco cede eree rend doe cheered ERR ER 91 Figure 30 Calling into a remote node from the PSTN 00 eee eee 97 Figure 31 MetMeeling options acoceueasscse sess ur9 Gard eee deeb X Doe eas 98 Figure 32 NetMeeting advanced options 0 2 00 cece eee es 99 Fui PO PANGSs occuciocalanen dees eins ds Moe rienes meee 4 PRX dU E a 102 Figure 34 Port ranges dialog DOK 2 006 c
29. Foy oe Rd or dor ac Rd oe he ek aid d dpa iod DA rcnt dh ER ERS d 26 DOS PON okerasebisepGISITRORESSSEITICIS edqqcS4P QI RU CEXQECiiQadxqeddelbeR 27 Chapter 2 Prerequisites checklist isirasaakRAaAwREARRARRAEAREARARARRRETA REA swede 29 goi d UTI au equa aep 54 AR XUPIRHRAOA REESE Aon XO lE ede pelo epp dd 29 guo OOUIEDS qoae S blame ded X Hb qo de eR qu iqqad b reque epp 30 Mebwork assessmelll x14 cote ei cde deeded goa ese ed WAGE ON eed XOU ACE eee a X Lr jd OR 30 Resource assessmMeoi Llicssexes DRE Ee Pod d Oe XA LACE kv Edo e an de ab d 31 Configuring media gateway parameters for IP service 0 000e eee aes 32 IP Telephony Configuration Guide 4 Table of Contents dioc P 3 Business Communications Manager system configuration llle 34 Defining published IP address 22s lues tows ox km Rx mE Rx Rx Xxx Ra 35 Setting the Global IP published IP 2i cca sc bees ERE RS Ex x RR E es 35 Determining the published IP address 0000 cece ee eee 36 IF telepDOFIB S os oos daos Sox sot RP sae daa denne Seance POP ORA ad ed doped ae AG eqs 37 Mervision wireless telephones iu cere ee ade eR Rode Ad xx daas Xo dea doge oa 37 Chapter 3 installing IP telephones ciasraasakrt 4a adaaRR REA EaRR RYRR PLA EARS AFER 39 Supporting IP telephony 2isesasecesis nemen RR be RR REC RS RR RR Ree E eS 39 About Nortel Networks IP telephones 0 2000 cece sels 40 Configuri
30. Guide for information about VLAN configuration and DHCP Also refer to Using VLAN on the network on page 148 When you have entered all the configuration information the telephone attempts to connect to the Business Communications Manager The message Locating Server appears on the display If the connection is successful the message changes to Connecting to Server after about 15 seconds Initialization may take several minutes Do not disturb the telephone during this time When the telephone connects to the server and is ready to use the display shows the time and date As well the six keys at the top of the display are labelled The telephone is ready to use Note If the DN record has not yet been configured as will be the case with auto assigned DNs you will only be able to make local calls until other lines have been assigned in the DN record Note If the telephone has not been registered before you will receive a New Set message Enter the information as prompted Refer to Registering the telephone to the system on page 45 IP Telephony Configuration Guide 48 Chapter3 Installing IP telephones Troubleshooting an IP telephone If the system is not properly configured several messages can appear Table 11 IP telephony display messages Message Description Solution SERVER NO PORTS LEFT The Business Communications Manager has run out of ports This message will remain on the disp
31. IP telephony traffic TCP traffic behavior Most of corporate intranet traffic is TCP based Different from UDP which has no flow control TCP uses a sliding window flow control mechanism Under this design TCP increases its window size increasing throughput until congestion occurs Congestion results in packet losses and when that occurs the throughput decreases and the whole cycle repeats When multiple TCP sessions flow over few congestion links in the intranet the flow control algorithm can cause TCP sessions in the network to decrease at the same time causing a periodic and synchronized surge and ebb in traffic flows WAN links can appear to be overloaded at one time and then followed by a period of under utilization There are two results e bad performance of WAN links e IP telephony traffic streams are unfairly affected P0993474 03 Quality of Service 161 Business Communications Manager router QoS support With a Business Communications Manager system the VoIP gateway and the router are in the same box The Business Communications Manager router performs QoS and priority queuing to support VoIP traffic The router supports VoIP in the following two ways e Ina DiffServ network the Business Communications Manager system acts as a DiffServ edge device and performs packet classification prioritization and marking The router performs admission control for H 323 flows based on the WAN link bandwidth and utilization
32. Public number field enter the DN Line 243 Received number Line 243 Received number Public number 2243 Target lines Line 241 Line 242 9 Line 243 General 9 Trunkline data The telephone assigned to that DN can now receive all calls with that DN number that come into the Business Communications Manager to which the telephone is connected P0993474 03 Chapter 5 Configuring VoIP trunks 91 For a detailed explanation about target lines see the Business Communications Manager 3 0 Programming Operations Guide Example configuration set to set This section walks through a sample Business Communications Manager configuration including e On Business Communications Manager Ottawa on page 92 e On Business Communications Manager Santa Clara on page 94 Making calls on page 95 e Connecting an i200X telephone on page 96 In this scenario shown in the following figure two Business Communications Managers in different cities are connected to a WAN One Business Communications Manager resides in Ottawa the other resides in Santa Clara Figure 29 Example PSTN fallback DN 3322 22244 Santa Clara Gateway 2 Gateway destination digit 2 Route 022 VoIP Route 222 PRI line Route 009 PSTN line with external 41613555 Dialout Destination code 2 6135552244 Route 022 absorb 0 Route 222 absorb All Destination code 9 Route 009 absorb 0 Route 00
33. When received the WAN link marks the H 323 flows as Premium traffic and places the flows in the high priority queue Note Differentiated Service DiffServ is a QoS framework standardized by the Internet Engineering Task Force IETF e Ina non DiffServ or a legacy network the router manages the WAN link to make sure Premium VoIP packets have high priority in both directions when crossing a slow WAN link Network Quality of Service This section discusses the quality of service aspects of networking e Network monitoring on page 162 e Quality of Service parameters on page 162 e Fallback to PSTN on page 163 Business Communications Manager VoIP Gateway uses a method like the ITU T Recommendation G 107 the E Model to determine the voice quality This model evaluates the end to end network transmission performance and outputs a scalar rating R for the network transmission quality The packet loss and latency of the end to end network determine R The model correlates the network objective measure R with the subjective QoS metric for voice quality MOS or the Mean Opinion Score This model provides an effective traffic building process by activating the Fallback to Circuit Switched Voice Facilities feature at call set up to avoid quality of service degradation New calls fall back when the configured MOS values for all codecs are below the threshold The model is the reason for compression character
34. a gatekeeper The Business Communications Manager can request a method for call signaling but whether this request is granted depends on the configuration of the gatekeeper Ultimately the gatekeeper decides which call signaling method to use To modify the settings for your local gateway 1 Inthe Unified Manager open Services IP Telephony and click on H 323 trunks The Local Gateway IP Interface screen appears P0993474 03 Chapter 5 Configuring VoIP trunks 105 Figure 35 Local gateway IP interface 138 52 hensive elete M 47 85 138 52 system Resources services Telephony Services IP Telephony Q System Configur Nortel IP Termin H 323 Terminals ee PortRanges Call Detail Recordin LAN CTE Configurat Cnnanle Servirte z r Local Gateway IP Interface Fallback to Circuit Switched Local Gateway IP Interface Media Parameters SCNFallback X Call Signaling CallSignaling v Gatekeeper IP Gatekeeper Alias Names AjasNames Registration TTL RegTTL Gateway Protocol G WProtocol v 2 Use the information in the table below to set up the Local Gateway IP interface record Table 19 Local Gateway IP interface fields Field Value Description Configuration note Refer to Using Radvision ECS 2 1 0 1 as the gatekeeper on page 108 and Using CSE 1000 as a gatekeeper on page 109 for
35. an unworkable feature on single line display sets including the M7100 and especially on Symbol Calls between Symbol sets do not support the Call Record feature There is sometimes significant echo heard on the Symbol set during ringback on outgoing calls over analog lines Business Communications Manager does not support remote registration for symbol sets if these sets are behind another device for example another Business Communications Manager or a third party router which has NAT turned on Each H323 Terminal configured utilizes one IP Client Resource whether the H323 Terminal is being used or not Firewall Default Rules when enabled block Symbol Registration and call processing You must add two additional rules 1 Pass Protocol TCP UDP Destination Port H 323 and 2 Pass Protocol UDP Destination port 1719 Ring cadence on Symbol handsets does not distinguish between Internal and External callers Symbol sets work fine as members of hunt groups but when they are answer DN twinned with other sets they do not ring under some circumstances When configured with an answer DN for a set in a hunt group Symbol sets sometimes do not ring or ring but do not display CLID information and cannot answer the incoming call It is recommended that the Symbol set be added to the hunt group before the answer DN set or that the Symbol set be designated as the prime DN with the answer DN for it applied to the twinned desk set This
36. bus 07 DS30 3 5 split as a processor for internal media traffic including IP telephony instead of for digital traffic through a media bay module IP Telephony Configuration Guide 166 Glossary enbloc All dialed digits sent in a single expression The system waits for all digits to be dialed before processing the call ESSID This is the code that identifies the access point that a NetVision handset uses to connect to the internet and the Business Communications Manager fallback to PSTN Your VoIP trunks can be configured to revert to land lines processed over the PSTN public switched telephony network if the IP network experiences quality issues This process occurs during call setup QoS must be active on the network to use this feature FEATURE 900 This feature code accesses a display menu on Nortel IP telephones You use the directional arrows on the telephone to access menu items which when selected perform as if you had entered that feature code This menu can also be accessed through the Services button default FEATURE 999 hot desking This feature allows you to transfer the telephone and call features temporarily from one IP telephone to another The originating IP telephone cannot be used during this period feature labels The names that appear beside the four six soft keys on Nortel IP telephones can be adjusted to better reflect local requirements Label changes are performed through the Unifie
37. can include a router that forms a connection to the Internet A Business Communications Manager can have up to two LAN connections Public Switched Telephone Network The Public Switched Telephone Network PSTN can play an important role in IP telephony communications In many installations the PSTN forms a fallback route If a call across a VoIP trunk does not have adequate voice quality the call can be routed across the PSTN instead either on public lines or on a dedicated ISDN connection between the two systems The Business Communications Manager also serves as a gateway to the PSTN for all voice traffic on the system Key IP telephony concepts In traditional telephony the voice path between two telephones is circuit switched This means that the analog or digital connection between the two telephones is dedicated to the call The voice quality is usually excellent since there is no other signal to interfere In IP telephony each IP telephone encodes the speech at the handset microphone into small data packets called frames The system sends the frames across the IP network to the other telephone where the frames are decoded and played at the handset receiver If some of the frames get lost while in transit or are delayed too long the receiving telephone experiences poor voice quality On a properly configured network voice quality should be consistent for all IP calls The following sections describe some of the components that det
38. channel is still equal to the full transmission rate The following figure shows the peak bandwidth requirements for one call on a full duplex link with silence compression enabled The spare bandwidth made available by silence compression is used for lower priority data applications that can tolerate increased delay and jitter Figure 50 One call on a full duplex link with silence compression Tx Hello Fred This is Susan Do you have a minute Rx Fred here Hi Sure Conversation di RE Bandwidth used Channel Link max BE ee Dime cS 5 x EA j Time C Am D ChannelLinkmax Bs c 2 Gg Y Y G Ti ais gt Independent Tx oad E bandwidth not shared by half duplex cds Bandwidth available for data apps IP Telephony Configuration Guide 140 Silence compression When several calls are made over a full duplex link all calls share the same transmit path and they share the same receive path Since the calls are independent the peak bandwidth must account for the possibility that all speakers at one end of the link may talk at the same time Therefore the peak bandwidth for n calls is n the full transmission rate The following figure shows the peak bandwidth requirements for two calls on a full duplex link with silence compression Note that the peak bandwidth is twice the full transmission rate even though the average bandwidth is considerably less The spare bandwidth made availab
39. circuit you selected highlighted On the Protocol menu click Add Select the protocol priority from the list Click the OK button Click Protocols Edit Protocol Priority and then click Priority Outbound Filters The Priority Outbound Filters window appears Click Template The Filter Template Management window appears Enter the template name and click Create The Create Priority Outbound Template window appears Type a descriptive name in the Filter Name field Click Criteria Add Datalink IP and then click Criterion The Add Range window appears If you choose the User Defined criterion the Add User Defined Field window appears first P0993474 03 Note The information in this section is not required for recent versions of the Nortel Networks routers such as BayRS release 15 that support prioritization based on the Interoperability 147 12 13 14 15 16 17 18 19 20 21 Type a minimum and maximum value to specify the range and then click the OK button The Add Range window closes The new criterion and ranges now appear in the Filter Information field of the Create Priority Outbound Template window Click Action Add and then click action Click the OK button The Filter Template Management window opens The new template appears in the templates list Click Done The Priority Outbound Filters window opens Click Create The Create Filter window opens Select a circuit in the Interf
40. does address most of the functionality problems There still appears to be a problem for calls routed by CCR 2 5 FP1 MR1 1 Gatekeeper Officially Business Communications Manager supports only ECS 2 1 0 1 gatekeeper Business Communications Manager does not support Call Setup Q 931 routing mode Business Communications Manager does not support the Radvision Dialing plan package ECS option Check that call is active every XXX seconds must be unchecked Radvision ECS 2 1 0 1 gatekeeper limitations ECS does not support fast start in the Call Setup Q 931 and Call Control H 245 routing mode IP Telephony Configuration Guide 152 Interoperability Table 36 Software interoperability restrictions and limitations Software release Description of restriction limitation 3 0 GA Gatekeeper e Officially Business Communications Manager supports RadVision ECS 2 1 0 1 and CSE 1000 as gatekeepers It does not support the Radvision Dialing plan package e Radvision ECS 2 1 0 1 gatekeeper limitations ECS does not support fast start in the Call Setup Q 931 and Call Control H 245 routing mode Call signaling By selecting GatekeeperRouted or GatekeeperResolved you switch Business Communications Manager to gatekeeper mode which means your Remote Gateway table will no longer be a part of your call routing plan Choosing one of the modes will advertise a Business Communications Manager prefere
41. gatekeeper you do not need to establish these gateways Refer to Using a gatekeeper on page 103 To add an entry to the remote gateway list 1 In Unified Manager open Services IP Telephony H 323 Trunks and click on Remote Gateway The remote gateway tab appears The Remote Gateway screen shows all gateway records that have been added to the system 2 On the top menu click Configuration and select Add entry If you are modifying an existing entry select the entry then under Configuration select Modify entry The Remote Gateway window appears as shown in the next figure P0993474 03 Chapter 5 Configuring VoIP trunks 79 Figure 20 Remote gateway dialog Remote Gateway Name Headotfice Format 15 characters Destination IP 000 000 000 000 QoS Monitor Disabled Y Transmit Threshold n T Receive Threshold lo uu EUST Gateway Type BCM 2 5 M Gateway Protocol None v Destination Digits 0000 3 Use the information in the table below to set up the remote gateway information Table 17 Remote gateway record Field Value Description Name lt alphanumeric gt Enter an indentifying tag for the remote system Destination IP lt ip address gt Enter the IP address of the remote system gateway QoS Monitor Disabled Choose Enabled if you intend to use a fallback PSTN line Ensure Enabled that QoS Monitor is also enabled on the remote
42. link utilization can be an average of a week a day or one hour To be consistent with the considerations get the peak utilization of the trunk 3 Determine the available spare capacity Business Communications Manager intranets are subject to capacity planning controls that ensure that capacity use remains below a determined utilization level For example a planning control can state that the utilization of a 56 kbit s link during the peak hour must not exceed 50 For a T1 link the threshold is higher at 85 The carrying capacity of the 56 kbit s link can be 28 kbit s and for the T1 1 3056 Mbit s In some IP Telephony Configuration Guide 122 Efficient Networking organizations the thresholds can be lower than those used in this example In the event of link failures spare capacity for rerouting traffic is required Some WAN links can exist on top of layer 2 services such as Frame Relay and Asynchronous Transfer Mode ATM The router to router link is a virtual circuit which is subject not only to a physical capacity limits but also to a logical capacity limit The installer or administrator needs to obtain the physical link capacity and the QoS parameters The important QoS parameters are CIR committed information rate for Frame Relay and MCR maximum cell rate for Asynchronous Transfer Mode ATM The difference between the current capacity and the acceptable limit is the available capacity For example a T1 link u
43. load splitting 134 in band signaling 143 Incoming call configuration 89 incremental IP telephony traffic 131 Installation 3 port switch 43 configuration display keys 45 12050 Software Phone 64 initialization IP telephones 47 IP telephone server parameters 46 IP telephones 39 NetVision telephones 67 NetVision before you start 69 post installation network measurements 134 restart to configure 45 Unified Manager configuration 53 Internet Control Message Protocol ICMP 141 Internet Engineering Task Force IETF internet 3 way switch 43 Interoperability 143 interoperability gatekeeper supports 109 intranet delay and error analysis 131 networking multiple Business Communications Manager Systems 117 other resource considerations 131 routing changes 134 WAN link resources 121 Invalid Server Address 48 50 IP address DHCP configuration 49 gatekeeper 104 H 323 terminals list 71 network prerequisites 29 networking 36 private 36 92 public 36 92 Published IP address 35 remote gateway 78 IP address conflict 48 IP datagram 141 IP packet 122 IP speech packets 75 IP telephones 3 port switch 43 before installation 43 block single telephone 49 codec jitter buffer settings 53 codecs 42 53 P0993474 03 Index 175 viewing 48 contrast level 49 defined 20 deleting handset record 72 deregister 62 deregistering online sets 62 DHCP 49 display keys for configuration 45 does not connect 48 ethernet connecti
44. ms for interoperability with the Business Communications Manager Maximum Delay Voice Playout Jitter Buffer digits Choose the minimum jitter buffer value you want to Nominal Delay allow Voice Playout Jitter Buffer digits Choose the maximum jitter buffer value you want to allow VAD checkbox enabled disabled gt Check or uncheck box to enable or disable silence suppression for the codec 8 Click Submit 9 Click Transfer for the node that you modified P0993474 03 Chapter 5 Configuring VoIP trunks 113 Gatekeeper call scenarios This section explains what must be set up and how a call would be processed for the two types of gatekeeper configurations The following figure shows a network with three Business Communications Managers and a gatekeeper Figure 36 Business Communications Manager systems with a gatekeeper gatekeeper E IP 10 10 10 17 Business Communications Manager Ottawa IP 10 10 10 18 Business Communications Manager Santa Clara IP 10 10 10 19 Business Communications Manager Calgary IP 10 10 10 20 This example explains how a call from DN 321 in Ottawa would be made to DN 421 in Santa Clara It assumes that call signaling is set to Gatekeeper Resolved and no pre granted ARQ has been issued 1 Business Communications Manager Ottawa sends an AdmissionRequest ARQ to the gatekeeper for DN 421 The gatekeeper resolves DN
45. of data services although most applications cannot identify a reduction of say 50 ms in delay Improvement of the network results in a network that is correctly planned for voice but over planned for data services Another plan is to consider using QoS in the intranet This provides a more cost effective solution to engineering the intranet for non homogenous traffic types IP Telephony Configuration Guide 160 Quality of Service Traffic mix This section describes QoS works with the IP telephony and what new intranet wide results can occur Before putting into operation QoS in the network determine the traffic mix of the network QoS depends on the process and ability to determine traffic by class so as to provide different services With an intranet designed only to deliver IP telephony traffic where all traffic flows are equal priority there is no need to consider QoS This network can have one class of traffic In most corporate environments the intranet supports data and other services When planning to provide voice services over the intranet the installer must determine the following e Is there existing QoS What kind IP telephony traffic must take advantage of established mechanisms if possible e What is the traffic mix If the IP telephony traffic is light compared to data traffic on the intranet then IP QoS can work If IP telephony traffic is heavy data services can be affected if QoS is biased toward
46. peak VoIP Load 8 28kbit s 224kbit s With Business Communications Manager VoIP gateway bandwidth requirements and Traceroute measurements the R4 R5 link is expected to support the Santa Clara Richardson Santa Clara Tokyo and the Ottawa Tokyo traffic flows The other IP telephony traffic flows do not route over R4 R5 A peak of eight calls can be made over RA R5 for the four IP telephony ports per site RA R5 needs to support the incremental bandwidth of 8 x 12 2 96 kbit s To complete this exercise the traffic flow from every site pair needs to be summed to calculate the load on each route and loaded to the link IP Telephony Configuration Guide 130 Efficient Networking Enough link capacity The following table sorts the computations so that for each link the available link capacity is compared against the additional IP telephony load For example on link R4 R5 there is capacity 568 kbit s to allow for the additional 96 kbit s of IP telephony traffic Table 31 Link capacity example Incremental IP telephony Link Utilization 96 load Available End Capacity capacity Traffic Enough Points kbit s Threshold Used kbit s Site pair kbit s capacity R1 R2 1536 85 75 154 Santa Clara 15 5 Yes Ottawa Santa Clara Tokyo R1 R3 1536 R2 R3 1536 R2 R4 1536 R4 R5 1536 85 48 568 Santa Clara 24 Yes Richardson Ottawa Tokyo Santa Clara Tokyo Some network mana
47. planned or damaged LAN segments Packets that arrive at the destination late are not placed in the jitter buffer and are lost packets See Adjust the jitter buffer size on page 133 IP Telephony Configuration Guide 134 Efficient Networking Routing issues Routing problems cause unnecessary delay Some routes are better than other routes The Traceroute program allows the user to detect routing anomalies and to correct these problems Possible high delay differences causes are routing instability e wrong load splitting frequent changes to the intranet e asymmetrical routing Post installation network measurements The network design process is continuous even after implementation of the IP telephony and commissioning of voice services over the network Network changes in regard to real IP telephony traffic general intranet traffic patterns network controls network topology user needs and networking technology can make a design invalid or non compliant with QoS objectives Review designs against prevailing and trended network conditions and traffic patterns every two to three weeks at the start and after that four times a year Ensure that you keep accurate records of settings and any network changes on an ongoing basis Ensure that you have valid processes to monitor analyze and perform design changes to the IP telephony and the corporate intranet These processes ensure that both networks continue to conf
48. pools to routes PSTN line pool to other system a b c On the navigation tree click the route you created for the PSTN line In the Use Pool box type the letter of the line pool for the fallback lines In the External field enter the dial numbers that access the other system through the PSTN For example 1 area code local code IP Telephony Configuration Guide 84 Chapter5 Configuring VoIP trunks PSTN line pool to local PSTN lines a On the navigation tree click the route you created for the PSTN line b In the Use Pool box type the letter of the line pool for the fallback lines C Leave the External field blank VoIP line pool a Onthe navigation tree click the route you created for the VoIP lines b Inthe Use Pool field type the letter of the line pool for the VoIP lines C Leave the External field blank unless the destination digit you entered for the remote gateway is different than the number you want to use for the destination code Creating destination codes for fallback Create a destination code that includes the VoIP and PSTN routes that you created in Configuring routes on page 82 to respond to the same access number destination code When this code is dialed the Business Communications Manager will select the VoIP line if possible If the line is not available the call will fall back to the PSTN line As well you need to create or ensure that your destination code 9 include
49. the Using a gatekeeper on page 103 The following table shows which networking applications are supported for each Business Communications Manager software release Table 37 Software network communications application compatibility Application compatibility BCM BCM BCM 2 5 2 5 FP1 Net ITG v version 2 03 2 5 FP1 MR1 Meeting X X Symbol GK CSE1K BCM 2 03 X basic call ITG v to from 25 24 BCM 2 5 X X basic call ITG v to from 25 25 BCM 2 5 X X X X ITG X FP1 25 25 FP1 MR 1 1 X X X X ITG X X 25 25 BCM 3 0 X X X ITG X X X 26 26 P0993474 03 153 Appendix E Quality of Service The users of corporate voice and data services expect these services to meet a level of quality of service QoS This in turn affects network design The purpose of planning is to design and allocate enough resources in the network to meet user needs QoS metrics or parameters help in meeting the needs required by the user of the service This section provides information about e Setting QoS on page 153 e Measuring Intranet QoS on page 155 Implementing QoS in IP networks on page 159 e Network Quality of Service on page 161 Setting QoS There are two interfaces that must be considered when you set up QoS on the network as shown in the figure below e P telephony interfaces with the end users voice services made available need to meet user QoS objectives
50. the dinfo command Example Enter show ip alerts routes italic text Represents terms book titles and variables in command syntax descriptions If a variable is two or more words the words are connected by an underscore Example The command syntax show at valid route valid route is one variable and you substitute one value for it bold text Represents fields names field entries and screen names in the Unified Manager application plain Courier Represents command syntax and system output such as prompts and text system messages Example Set Trap Monitor Filters Acronyms This guide uses the following acronyms ATM BCM CIR DID DOD DIBTS DSB Asynchronous Transfer Mode Business Communications Manager Committed Information Rate Direct Inward Dialing Direct Outward Dialing Digital In Band Trunk Signalling DIBTS Signalling Buffer P0993474 03 Preface 15 DSL ICMP IEEE802 ESS ITG ITU IXC IP ISDN LAN LATA LEC MOS NVPA PCM PING PiPP PPP PRI PSTN QoS RAS RTP SNMP TCP UDP UTPS VoIP VAD VLAN WAN Digital Subscriber Line Internet Control Message Protocol Institute of Electrical and Electronics Engineers Inc standard 802 Electronic Switching System Identification code Internet Telephony Gateway for Meridian International Telecommunication Union IntereXchange Carrier Internet Protocol Integrated Services Digital Network Local Area Network Local Access and Transport A
51. the network diagram contain IP Addresses netmasks and network locations of all Business Communications Managers 1 d Answer this if your system will use IP trunks otherwise leave it blank Does the network diagram contain IP Addresses and netmasks of any other VoIP gateways that you need to connect to 1 e Answer this only if your system will use a gatekeeper otherwise leave it blank Does the network diagram contain the IP address for any Gatekeeper that may be used IP Telephony Configuration Guide 30 Chapter2 Prerequisites checklist Network devices The following table contains questions about devices on the network such as firewalls NAT devices and DHCP servers e If the network uses public IP addresses complete 2 d e Ifthe network uses private IP addresses complete 2 e to 2 f Table 2 Network device checklist Prerequisites Yes No 2 a Is the network using DHCP 2 b If so are you using the DHCP server on the Business Communications Manager 2 c Is the network using private IP addresses 2 d Are there enough public IP addresses to accommodate all IP telephones and the Business Communications Manager 2 e Does the system have a firewall NAT device or will the Business Communications Manager be used as a firewall NAT device NOTE NetVision handsets do not work on a network that has NAT between the handset and the system 2 f If the Business Communicat
52. the specific Business Communications Manager details the IP telephone will automatically attempt to find the server Refer to Configuring DHCP on page 49 which describes the secific DHCP requirements for IP telephones and to the Programming Operations Guide which provides detailed DHCP configuration information Once you register the telephone to the system as described in Registering the telephone to the system the telephone assumes the parameters it receives from the system which are described in Configuring telephone settings If DHCP is not configured to provide system information or if you are not using DHCP on your network you need to configure your telephone parameters before the telephone can register to the system In this case follow the directions in Configuring telephone settings and then follow any of the prompts that appear as described in Registering the telephone to the system P0993474 03 Chapter 3 Installing IP telephones 45 Registering the telephone to the system When you first connect the telephone to the IP connection you may receive one of the following e If the telephone is not yet registered and if a password was entered in the Terminal Registration screen the telephone prompts you for that password e If you set Auto Assign DN on the Business Communications Manager to OFF the telephone prompts you for a DN e If you are prompted for a password enter the password and press OK
53. wens E odeeeee as 162 Poeran 23a bap uaodoo diss Scenes bd E D dead iv diu acd e du diua deans 162 Delay variation tier 2 cccceescdnewcdans TERRE REOR oe eee EORR RR Ea HR RR 163 Fallacies quisi es Sogl va Ea apod iR RT 163 ILI la M EPETETESRT IET DUI C Lo t LII SIS 165 NAEK ioasayARWAAXARRAAAEAREAARAAAARRARERRAAANARAWEAMTEAAAAAARARP EE 171 P0993474 03 Figures Figure 1 NIE CIARA 6 8d ah ane ndaGe Soe hE ROS A RC XAdQXd RC G Ggqadccas 22 Figure2 System Configuration Parameters screen 000 c eee eee 32 Pere obel IP clings cosa eda ne De eR he a Wel ee 35 Figure4 Setting the Published IP address 0 0202000 RR hmm 36 Figure5 Set registration properties naana nunana naaa 41 Figure6 IP terminal registration server status seien 52 Figure IP Terminal Stas suceso mace e keai i intenet En EEr 53 Figure 8 IP Terminal status dialog sss eder cR Eo URREA RPG REA x3w Rd 54 Figure9 IP Telephony Features LISt issus Ree eR RR RE ERE ES E RR 55 Figure 10 Add Modify Telephony Features List 0 0 cece ee eee 56 Figure 11 IP Terminal Status tab list iius cod iex uke eke Conde a be 57 Figure 12 Label ser Css lt cavaieuen d004 Some nw adque d bea sque a dq pes 59 Figure 13 Deregister DN from Configuration menu 2 0 ee eee 62 Figure 14 12050 Communications STVBF oa eke ak Rx Re A ACC 64 Figure 15 12050 Switch type isocusessexessexeraekzesede rediri ee RR ex Re ORO T 65
54. will determine which system features are available to the caller Configuring a VoIP trunk requires the following actions e Pre installation system requirements on page 74 e Configuring media parameters on page 74 e Outgoing call configuration on page 76 e Incoming call configuration on page 89 Note If you are using the Business Communications Manager with a Meridian 1 M1 ITG system you must set up the system to be compatible with the M1 Refer to Interoperability on page 143 e This section also includes information about Example configuration set to set on page 91 e Remote access over VoIP trunks on page 97 e Configuring Net Meeting clients on page 98 e Quality of Service Monitor on page 100 e Port settings on page 101 e Using a gatekeeper on page 103 Note Also refer to Appendix D Interoperability on page 143 for information about interoperability issues IP Telephony Configuration Guide 74 Chapter5 Configuring VoIP trunks Pre installation system requirements Ensure that you have obtained the following before continuing Keycodes Before you can use VoIP you must obtain and install the necessary keycodes See the Keycode Installation Guide for more information about installing the keycodes Talk to your Business Communications Manager sales agent if you need to purchase VoIP keycodes Each keycode adds a specific number of VoIP trunks You mu
55. will either get a busy signal or be rerouted to the Prime set depending on how your system is programmed The same type of delay occurs when the IP telephone is reconnected to the system IP Telephony Configuration Guide 64 Chapter3 Installing IP telephones Configuring the Nortel Networks i2050 Software Phone The Nortel Networks 12050 Software Phone allows you to use a computer equipped with a sound card microphone and USB headset to function as an IP terminal on the Business Communications Manager system The Nortel Networks 12050 Software Phone uses the computer IP network connection to connect to the Business Communications Manager The registration process is the same as for the 12002 and 12004 telephones Registering the telephone to the system on page 45 When you install the Nortel Networks 12050 Software Phone on screen documentation walks you through the steps for installing the software You can also refer to the i2050 Software Phone Installation Guide To configure the Nortel Networks 12050 Software Phone to connect to the Business Communications Manager 1 Click the Start button and then click Settings 2 Click Control Panel 3 Double click the 12050 Software Phone icon The utility opens to the Communications Server tab Figure 14 i2050 Communications server i2050 Software Phone Properties E x Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound De
56. 0 gateway Destination Digits lt numeric gt Set the leading digits which callers can dial to route calls through the remote gateway Ensure that there are no other remote gateways currently using this combination of destination digits If multiple leading digits map to the same remote gateway separate them with a space For example 7 81 9555 These numbers are passed to the remote system as part of the dialed number 4 Click Save Configuring PSTN fallback By enabling PSTN fallback you allow the system to check the availability of suitable bandwidth for a VoIP call then switch the call to a land line if the IP line is not available or cannot produce the expected quality The following figure shows how a fallback network would be set up between two sites Figure 21 PSTN fallback diagram Business Communications Manager A Business Communication Manager B PSTN P0993474 03 Chapter 5 Configuring VoIP trunks 81 In a network configured for PSTN fallback there are two connections between a Business Communications Manager and a remote system One connection is a VoIP trunk connection through the IP network The fallback line is a PSTN line which can be the public lines or a dedicated T1 BRI PRI or analog line E amp M to the other system When a user dials the destination code the system checks first to see if the connection between the two systems can support an appropriate l
57. 000 codec compatibility with endpoints Business Communications Manager preferred codec Refer to Configuring media parameters on page 74 CSE 1000 codec configuration silence suppression is disabled G 711 ulaw or G 711 alaw silence suppression has no effect G 729 G 729 AB is enabled silence suppression is enabled G 729A and G 723 are disabled G 729 G 729A is enabled silence suppression is disabled G 729AB and G 723 are disabled G 723 Not supported on CSE 1000 silence suppression is enabled G 723 G 723 is enabled G 729A and G 729AB are disabled G 711 is always part of the CSE 1000 configuration and cannot be removed Setting Codecs on the CSE 1000 Use the Element Manager tool to set the codec information for the CSE 1000 This tool can be accessed at http lt SignalingServerIP gt In the tool select Configuration Select IP Telephony Click on DSP Profile a fF WON name In the Node Summary Window select the node to be configured and click on Edit On the list of codecs enable or disable each by clicking on the check box beside the codec To view or change the codec configuration click on the codec name 7 Ensure the following fields are set Table 27 CSE 1000 codec configuration Field Value Description Codec Name codec name gt Name of the codec you selected Voice Payload Size msec per frame Choose the payload size for the codec Use 30
58. 111 are ASCII encoded decimal numbers representing the IP address of the Business Communications Manager They do not need to be three digits long as the and delimiters will guarantee parsing For example 001 01 and 1 would all be parsed correctly and interpreted as value 0x01 internal to the 12004 Note that these fields must be no more than three digits long each ppppristhe port number in ASCII encoded decimal It does not need to be five digits long as the and delimiters will guarantee parsing For example 05001 5001 1 00001 etc would all be parsed correctly and accepted as correct The valid range is 0 65535 stored internally in 12004 as hexadecimal in range 0 0xFFFF Note that this field must be no more than five digits long e inall cases the ASCII encoded numbers are treated as decimal values and leading zeros are ignored More specifically a leading zero does not change the interpretation of the value to be OCTAL encoded For example 0021 021 and 21 are all parsed and interpreted as decimal 21 Checking IP server status You can perform a status check on the Business Communications Manager server that gets used to register IP terminals 1 Inthe Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary screen appears Figure 6 IP terminal registration server status Configuration Performance Fault Report Tools Logoff View Help Summary r Summary
59. 3 Good Good Richardson Santa Clara Ottawa Santa Clara Tokyo Richardson Ottawa Richardson Tokyo Ottawa Tokyo Other measurement considerations The Ping statistics described above measure the intranet before IP telephony installation The measurement does not take into consideration the expected load provided by the IP telephony users If the intranet capacity is tight and the IP telephony traffic is important the installer or administrator must consider making intranet measurements under load Apply load using traffic generator tools The amount of load must match the IP telephony offered traffic estimated in the Business Communications Manager VoIP Gateway Bandwidth requirements P0993474 03 Quality of Service 159 Decision does the intranet meet IP telephony QoS needs The end of the measurement and analysis is a good indicator of whether the corporate intranet can deliver acceptable voice and fax services The Expected QoS level column in the table under Measurement procedure on page 158 indicates to the installer or administrator the QoS level for each site pair with the data To provide voice and fax services over the intranet keep the network within a Good or Excellent QoS level at the Mean o operating area Fax services must not travel on routes that have Fair or Poor QoS levels If QoS levels of some or all routes fall short of being Good evaluate options and
60. 323 third party devices through this type of connection Gatekeepers on the network A gatekeeper tracks IP addresses of specified devices and provides authorization for making and accepting calls for these devices A gatekeeper is not required as part of the network to which your Business Communications Manager 3 0 system is attached but Gatekeepers can be useful on networks with a large number of devices Referring to Figure 1 on page 22 for example Digital telephone A wants to call IP telephone B which is attached to Business Communications Manager B over a network that is under the control of a gatekeeper Digital telephone A sends a request to the gatekeeper The gatekeeper depending on how it is programmed provides Digital telephone A with the information it needs to contact BCM B over the network BCM B then passes the call to IP telephone B Note The Business Communications Manager does not contain a gatekeeper application If you want to put a gatekeeper on your network it must be put on a separate gatekeeper server The Business Communications Manager is compatible with RadVision and CSE 1000 gatekeepers Refer to Appendix D Interoperability on page 143 IP network In the network shown in Figure 1 on page 22 several LANs and a WAN are shown When planning your network be sure to consider all requirements for a data network Your network administrator should be able to advise you about the network setup and how the
61. 67 23 167 23 30 163 2 163 23 163 23 150 4 150 43 150 43 1542 154 23 154 23 G 729 10 121 6 60 8 60 8 83 2 41 6 41 6 94 4 47 2 47 2 8 kb s 20 68 8 34 4 34 4 49 6 24 8 24 8 55 2 27 6 27 6 30 51 2 25 6 25 6 38 4 19 2 19 2 42 2 21 1 21 1 G 723 1 30 48 0 24 0 24 0 35 2 17 6 17 6 39 0 19 5 19 5 6 3 kb s G 723 1 30 45 8 22 9 22 9 33 0 16 5 16 5 36 8 18 4 18 4 5 3 kb s Notes 1 Gray background indicates payload sizes used by Business Communications Manager 2 5 for transmission Other values listed indicate payload sizes that BCM can receive 2 Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter frame gap 3 G 711 does not support silence suppression With no silence suppression both the transmit path and the receive path continuously transmit voice packets Therefore the peak bandwidth requirement per call on half duplex links is Half Duplex links No Silence Suppression Peak Bandwidth per call 2 2 Continuous Transmission Rate On half duplex links with silence suppression enabled the half duplex nature of normal voice calls allows the sender and receiver to share the same bandwidth on the common channel While the sender is talking the receiver is quiet Since only one party is transmitting at a time silence suppression reduces the peak bandwidth requirement per call on a half duplex link to Peak Bandwidth per call 1 Continuous Transmission Rate Half Duplex links With Silence S
62. 800 800 89009 Fax 44 191 555 7980 email emeahelp Q nortelnetworks com CALA Caribbean amp Latin America Technical Support CTAS Telephone 1 954 858 7777 email csrmgmt nortelnetworks com P0993474 03 Preface 17 APAC Asia Pacific Technical Support CTAS Telephone 61 388664627 Fax 61 388664644 email asia support nortelnetworks com IP Telephony Configuration Guide 18 Preface P0993474 03 19 Chapter 1 Introduction IP Telephony provides the flexibility affordability and expandability of the Internet to the world of voice communications This section includes an overview of the components that make up the Business Communications Manager version 3 0 IP telephony and Voice over IP VoIP features e IP telephones and VoIP trunks on page 20 e Creating the IP telephony network on page 21 e Key IP telephony concepts on page 25 Business Communications Manager 3 0 with voice over IP VoIP provides several critical advantages e Cost Savings IP networks can be significantly less expensive to operate and maintain than traditional networks The simplified network infrastructure of an Internet Telephony solution cuts costs by connecting IP telephones over your LAN and eliminates the need for dual cabling Internet Telephony can also eliminate toll charges on site to site calls by using your existing WAN By using the extra bandwidth on your WAN for IP Telephony
63. 9 absorb 0 Dialout IP network Packet Data Network Dialout 14085553322 DN 2244 Ottawa Gateway 3 Gateway destination digit 3 Route 033 VoIP Route 333 PRI line Route 009 PSTN line with external 1408555 Destination code 3 Route 033 absorb 0 Route 333 absorb All Destination code 9 Route 009 absorb 0 Route 009 absorb 0 The systems already communicate through a PRI line which will be configured to be used for fallback Both systems already have all keycodes installed for eight VoIP lines and resources properly allocated for VoIP trunking For information about keycodes see the Business Communications Manager Keycode Installation Guide For information about Resource Allocation see Configuring the MSC Resources in the Business Communications Manager Programming Operations Guide IP Telephony Configuration Guide 92 Chapter5 Configuring VoIP trunks Each Business Communications Manager has 10 telephones that will be using VoIP lines In this setup only eight calls can be sent or received at one time If all 10 telephones attempt to call at the same time two of the calls will be rerouted to the PSTN Business Communications Manager Ottawa Business Communications Manager Santa Clara e Private IP address 10 10 4 1 e Private IP address 10 10 5 1 e Public IP address 47 62 54 1 e Public IP address 47 62 84 1 e DNs 2000 2999 e DNs 3000 3999 e From this system dial
64. 9 to get onto PSTN e From this system dial 9 to get onto PSTN On Business Communications Manager Ottawa This procedure details actions that the installer performs to set up the Business Communications Manager Ottawa 1 The installer sets up 2221 as the Control set for each VoIP line so that the VoIP schedule can be manually activated This setup is necessary for PSTN fallback The installer sets the published IP address In this case the public IP network is connected to the LAN 2 connection therefore the installer sets the published IP address to LAN 2 This is the address that devices on the Packet Data Network PDN will use to locate the system The installer configures the media for the system using the following settings e The first preferred codec is set to G 729 The installer chooses this setting due to the unique requirements of this installation e Silence Compression is turned on e Jitter Buffer is set to medium The installer puts eight VoIP lines into line pool O Any line pool can be used as long as all of the lines in the pool are VoIP trunks The installer does not set an access code for the line pool because the access code does not work with fallback Instead the line pool will be accessed using destination digits after the installer sets up PSTN fallback For each telephone on the system the installer gives the DN record access to line pool O The installer sets up a remote gateway for the Sa
65. 993474 03 Chapter 3 Installing IP telephones 55 Table 13 IP Terminal Status fields Continued Field Value Description JitterBuffer Auto Increase the jitter buffer size for any telephone that has poor network Default connectivity to the Business Communications Manager None Refer to Choosing a Jitter Buffer on page 43 Small Medium Large Terminal ID Read only 7 Click Save Working with the features list You can add and modify the features that display on the IP telephone feature list which is accessed through the Services button or by using FEATURE 900 Refer to Using the Services button to access features on page 56 Note that the list assigns the hot desking feature to position 1 refer to Using the Hot Desking feature on page 57 The Programming Operations Guide provides a complete list of Business Communications Manager Features and index codes 1 Inthe Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary appears 2 Click on the Telephony Features list tab Figure 9 IP Telephony Features List Telephony Features List uuu uc uM INE I o 00 o eoe oo e UU RAMNERNMDEE A oie Fi F2 F3 F4 F5 F6 F7 F8 F9 F10 F11 F12 F13 Hot desking 999 Last Number Redial 5 Conference 3 Do Not Disturb 85 Call Forward 4 Page 60 Background Music 86 Call Park 74 Call Pickup 75 Voice Call 66 Speed Dial 0 Message Send 1 SWCA 1 521
66. AN and it also provides the VLAN information for the network Refer to the Programming Operations Guide for the DHCP settings for VLAN Refer to Configuring the 12002 or 12004 telephone to the system on page 44 for information about configuring VLAN on the 12002 or 12004 telephone P0993474 03 Interoperability 149 Assigning VLANs becomes important if you have multiple devices connected to the same switch port such as when you use a 3 port switch to connect a computer and IP phone on the same network cable In this case the system needs to apply the correct VLAN for each device Specifying the site specific options for VLAN The Business Communications Manager DHCP server resides in default VLAN and is configured to supply the VLAN information to the IP phones The DHCP server will supply site specific option in the DHCP offer message The following definition describes the Nortel 12004 specific Site Specific option This option uses the reserved for site specific use DHCP options DHCP option values 128 to 254 and must be returned by the DHCP server as part of each DHCP OFFER and ACK message for the 12004 to accept these messages as valid The 12004 will pull the relevant information out of this option and use it to configure the IP phone Format of field is Type Length Data Type 1 octet Five choices 0x80 0x90 0x9d Oxbf Oxfb 128 144 157 191 251 Providing a choice of five types allows the 12004 to work i
67. Business Communications Manager fits into the network WAN A Wide Area Network WAN is a communications network that covers a wide geographic area such as state or country For Business Communications Manager 3 0 a WAN is any IP network connected to a WAN card on the Business Communications Manager 3 0 system This may also be a direct connection to another Business Communications Manager 3 0 system If you want to deploy IP telephones or NetVision telephones that will be connected to a LAN outside of the LAN that the Business Communications Manager is installed on you must ensure the Business Communications Manager has a WAN connection This includes ensuring that you obtain IP addresses and routing information that allows the remote telephones to find the Business Communications Manager and vice versa P0993474 03 Chapter 1 Introduction 25 The Business Communications Manager 3 0 Programming Operations Guide has a data section that describes the internet protocols and data settings that the Business Communications Manager requires or is compatible with Ensure that this connection is correctly set up and working before you attempt to deploy any remote IP devices LAN A Local Area Network LAN is a communications network that serves users within a confined geographical area For Business Communications Manager 3 0 a LAN is any IP network connected to a LAN card on the Business Communications Manager 3 0 system Often the LAN
68. Communications Manager you can delete the DN record for the terminal 1 Inthe Unified Manager open Services IP Telephony and click on H 323 Terminals 2 On the IP Terminal Status screen select the terminal you want to change 3 Inthe Configuration menu click Delete Entry A query box appears Select an Option x Are you sure you want to delete this row Yes No Cancel Waring Applet Window 4 Click Yes to delete the record Under the Systems DNs heading the DN record returns to the Inactive DNs list P0993474 03 73 Chapter 5 Configuring VoIP trunks This section explains how to configure voice over IP VoIP trunks on a Business Communications Manager 3 0 system A VoIP trunk allows you to establish communications between a Business Communications Manager and a remote system across an IP network Note VoIP trunks can be used for calls originating from any type of telephone within the Business Communications Manager system Calls coming into the system over VoIP trunks from other systems can be directed to any type of telephone within the system You cannot program DISA for voice over IP VoIP trunks therefore you cannot use COS passwords to remotely access features over your system The exception to this would be a tandem system where the call comes into system A over the PSTN then tandems to system B over an VoIP trunk In this case the remote access package set up for the COS password
69. DNs depending on the incoming digits This process is independent of the trunk over which the call comes in Other options You can assign the target line to a number of telephones if you want the call to be answerable to a call group for instance If System Wide Call Appearance SWCA keys are configured on memory buttons on the telephones the incoming line acts the same way as any other incoming call which depends on how SWCA has been set up to behave Refer to the Business Communications Manager 3 0 Programming Operations Guide and the Telephony Feature Handbook for more information about setting up SWCA keys You can assign the target line number to a Hunt Group DN if you want the call to appear on a group of telephones set up as a hunt group Refer to the Business Communications Manager 3 0 Programming Operations Guide for more information about setting up Hunt groups Mapping target lines involves two steps The target line is mapped to a telephone or Hunt group by assigning a free target line 241 to 492 to the telephone or Hunt group DN record The incoming digits e g 3321 are mapped to a target line the same one you assigned to the telephone by setting the Received Number under that target line to the incoming digits If your system does not have target lines already assigned use this procedure to assign target lines to individual telephones Note You can also use the Add Users wizard if you need to create tar
70. E Evening In this case the user dials the destination code plus the DN The destination code is absorbed but the system dials out the access number 1 XXX XXX before the DN digits Note This same process will be necessary if you are part of a Universal Dialing Plan UDP where each system is assigned a private access code that is not part of the DN and you want your users to be able to just dial the DN of the telephone they are calling In that case you enter the private access code in the External field and that gets dialed out before the DN 8 Repeat these steps for your destination code 9 a Under the destination code select the Normal schedule b Specify the route you created for the local PSTN C Set the absorb length to 0 d Repeat these steps for the VoIP schedule P0993474 03 Chapter 5 Configuring VoIP trunks 87 Activating the VoIP schedule Before activating the VoIP schedule calls using the destination code are routed over the PSTN This is because the system is set to use the Normal schedule which routes the call over the PSTN Once the VoIP schedule is activated calls made with the VoIP destination code are routed over the VoIP trunk The VoIP line must be activated from the control set for the VoIP trunk which is specified when the trunk is created Services Telephony Services Lines VoIP lines Enabled VoIP lines Line XXX General For information about control sets and configuring VoIP line
71. E EROR A OH 100 Table 19 Local Gateway IP interface fields 000000 ee eee eee 105 Table 20 Route and Dialing Plan configurations for NPI TON 107 Table 21 Radvision Calls screen required settings llle 108 Table 22 Radvision Advanced screen required settings 2 eee eae 108 Table 23 Radvision Predefined Endpoints Properties settings 109 Table 24 CSE 1000 H 323 endpoints siscssosabu eR RARE RR nx RR eR RR 110 Table 25 CSE 1000 H 323 dialing pali 224239 9 oe ERR sodas dob aa Eae ded 111 Table 26 CSE1000 codec compatibility with endpoints leeren 112 Table27 GCSE TODO codec configuration iix Lese cux o sp xr dsrin dee xem doa 112 Table 28 VolP Transmission Characteristics for unidirectional continuous media stream 122 Table 29 Bandwidth Requirements per Gateway port for half duplex links 124 Table 30 Bandwidth Requirements per Gateway port for Full duplex links 125 Table 31 Link capacity exampla ccc eye ed eee cee e ieee RERO EX E Race RR 130 Table 32 Business Communications Manager 3 0 Product Interoperability Summary 143 Table 33 Engineering specifications csaeaesuue cake ERO GE EG REG X GER d RE 143 Table 34 Supported voice payload sizes 00 000s 144 Table 35 NemSCONPANSON uaa lt sidsoncravka cise deed isd ode DA 145 Table 36 Software interoperability restrictions and limitations 150 T
72. IP traffic to other compatible systems with VoIP trunks Both digital and IP telephones can use these channels P0993474 03 171 Index Symbols amp nopage Voice Activity Detection see VAD 135 Numbers 3 port switch IP telephones 43 relocating IP telephones 63 A absorbed length 85 access code line pool 76 network example 92 Unified Manager programming 77 acronyms 14 active calls deregistering disruption 62 Address Range IP telephones 50 a law 145 Alias Names Local Gateway 106 Aliases Radvision 109 Allowed Services Radvision 109 assessment network 30 resources prerequisite 31 asymmetrical media channel negotiation 145 routing 134 Asynchronous Transfer Mode ATM 122 background noise 140 bandwidth available for other data 140 characteristics 122 determining requirements 121 full duplex links 125 half duplex link silence suppression 124 half duplex links 124 peak 123 silence compression 135 spare bandwidth 122 before you start IP telephony and network prerequisites 29 NetVision 69 block IP telephone dialout 49 bottlenecks 131 bridges network prerequisites 29 buffer jitter 43 buffers VoIP trunks 76 Business Communications Manager call chain network configuration 119 connecting to remote IP telephones 120 gateway router support 161 H 323 gateway specifications 143 MCDN system requirements 116 network device prerequisites 30 networking multi locations with call center 119 n
73. Name UTPS Status Up Version 30 100 90 12 Description UNISTIM Terminal Proxy Server P0993474 03 Chapter 3 Installing IP telephones 53 The following fields provide information about the IP server Field Value Description Name UTPS Name of the server Status Up UP server is operating Enabled Enabled Server is using DHCP Disabled Disabled server is not working Version read only current version of server software Description read only description of server Modifying IP telephone status settings Settings such as jitter buffers and codecs for the Nortel IP telephones including the 12050 12002 and 12004 can be modified through the Unified Manager 1 In the Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary appears 2 Click on the IP Terminal Status tab Figure 7 Edit Configuration Performance On the IP Terminal status screen every IP telephone currently connected to the Business Communications Manager occupies a row in the IP Terminal Status table as shown in the figure below IP Terminal status General IP Terminal Status I Status Fault Report Tools Logoff View Help ELE p MM 2431 Offline i2050 Default N Default 2432 Oie i2004 N24 Default IN A Default N A 2433 Offline i2002 N Default INVA Default N A 3 Select the IP Terminal that you want to cha
74. P telephones Table 8 IP telephone provisioning Prerequisites Yes No 7 a Are IP connections and IP addresses available for all IP telephones made available for telephone installers Hint Use the Programming Record form 7 b If DHCP is not being used has all telephone configuration been documented and port number been supplied if one is being used on the switch 7 c If DHCP is not being used or if you want to enter the port manually has the VLAN 7 d Have telephone power and connectors been provisioned minimum system requirements including headset 7 e Do computers that will be using the Nortel Networks i2050 Software Phone meet the NetVision wireless telephones Refer to Gathering system information before you start on page 69 IP Telephony Configuration Guide 38 Chapter2 Prerequisites checklist P0993474 03 39 Chapter 3 Installing IP telephones An IP telephone converts the voice signal into data packets and sends these packets directly to another IP telephone or to the Business Communications Manager over the LAN or the internet If the destination is an IP telephone the arriving voice packets are converted to a voice stream and routed to the speaker or headset of the target telephone If the destination is the Business Communications Manager the voice stream is routed to a circuit switched connection such as a telephone internal or line external
75. P0993474 03 Chapter 3 Installing IP telephones 59 Refer to the Telephony Feature Handbook for details about using this feature Customizing feature labels When your IP telephone acquires a DN record the default settings are applied to the telephone including assigning features to the memory keys on the telephone These features all have pre defined labels and the telephone automatically displays the appropriate labels beside the programmed buttons If you want to customize these labels to be more appropriate you can do so through the Feature Labels heading on the Unified Manager The screens under the Feature Labels heading allow you to define custom labels for 24 features The system comes with 10 default labels which are feature and language specific depending to which region your system was assigned The default labels are mainly messaging and call attendant features However you can change any other feature label by adding to this list or by deleting any of the default settings and inserting new codes and labels Follow these steps to change the features or labels on the memory buttons on your IP telephone 1 Click on the keys beside Telephony Services General Nortel IP terminals and Feature labels 2 Click on the label set you want to view The Labels label number screen appears Figure 12 Label set defaults Labels 7 12 Del All Labels 7 12 15 138 52 Feature 382 es Label Mail
76. PSTN or private network or some form of gateway VoIP Note IP telephones require an IP network to reach the Business Communications Manager However they do not need to use VoIP trunks to communicate beyond the Business Communications Manager They can use any type of trunk in the same way that digital telephones do Before setting up IP clients you must enable keycodes for IP telephony For information about entering IP Client keycodes see the Keycode Installation Guide Each IP Client keycode opens a specific number of IP telephone channels on the system Channels are distributed on a one to one basis as each IP telephone or NetVision handset registers with the system This section includes information about e Supporting IP telephony on page 39 e Configuring Nortel Networks i series telephones on page 40 e Modifying IP telephone status settings on page 53 e Working with the features list on page 55 e Download firmware to a Nortel IP telephone on page 60 e Deregistering DNs for IP telephones on page 62 e Moving IP telephones on page 63 e Configuring the Nortel Networks 12050 Software Phone on page 64 Supporting IP telephony The Business Communications Manager supports two types of IP telephony protocols UNISTIM and H 323 e The Nortel Networks i series telephones use the UNISTIM protocol e The Symbol NetVision and NetVision Data telephones use H 323 Refer to Chapter 4 Installin
77. Part No P0993474 03 Business Communications Manager 3 0 IP Telephony Configuration Guide NORTEL NETWORKS Copyright 2002 Nortel Networks All rights reserved The information in this document is subject to change without notice The statements configurations technical data and recommendations in this document are believed to be accurate and reliable but are presented without express or implied warranty Users must take full responsibility for their applications of any products specified in this document The information in this document is proprietary to Nortel Networks NA Inc Trademarks NORTEL NETWORKS is a trademark of Nortel Networks Microsoft MS MS DOS Windows and Windows NT are registered trademarks of Microsoft Corporation Symbol Spectrum24 and NetVision are registered trademarks of Symbol Technologies Inc All other trademarks and registered trademarks are the property of their respective owners P0993474 03 Table of Contents al rl Pep A 13 Before you Degli ceci nacen ku eios RR ERR REPE EAS EX RE ORE DDR eee eo ed 13 Symbols used in this guid caos uit dr ga ee edid e ebd e aped us scel rc ose 13 TOONE du ccdaeense shoes hdd u nee say db4chesaiaesekoss sows denes Sonus 14 un rem 14 Rolated pHDIGBBOPE ona eke he au coh ae dope qcietdr acies ded dad cba dca de ade se gd 16 How io get help uec poten 9 aree d kb CRGA o FRE Red d CEPR Ee CER EUR deme 16 Bore C
78. RI lines You still require the MCDN and IP telephony software keys and compatible dialing plans on all networked systems The one difference between MCDN over PRI and MCDN over VolP is that the VoIP trunks require specific Remote Gateway settings Under Services IP Telephony H 323 Trunks Remote Gateway ensure that Gateway Protocol is set to SL 1 for the VoIP connection to the Meridian system The Gateway Type would be set to ITG M1 Internet Telephony Gateway as it would for any non MCDN VolP connection to a Meridian system For details about setting up MCDN networks refer to the Private Networking chapter in the Business Communications Manager 3 0 Programming Operations Guide Note If you use MCDN over VolP ensure that your fallback line is a PRI SL 1 line to maintain MCDN features on the network One application of this type of network might be for a company which has an M1 at Head Office who want to set up a warehouse in another region This would allow the warehouse to call Head Office across VoIP lines bypassing long distance tolls This type of network also provides the possibility of having common voicemail off the M1 Refer to the following figure for an example IP Telephony Configuration Guide 116 Chapter6 Typical network applications using MCDN Figure 37 M1 to Business Communications Manager network diagram Meridian m eet Company server Head Office Warehouse M1 Business Co
79. RVER UNREACHABLE RESTARTING 48 service setting manual 82 Services key feature 900 menu list 56 Set IP 46 signaling method 104 silence compression 143 about 135 comfort noise 140 full duplex 138 half duplex 136 silence suppression full duplex links 125 half duplex links 124 site pairs 156 SL 1 MCDN fallback 116 MCDN over VoIP 80 SLI Gateway Protocol 107 slow connection IP telephones 48 Sniffer 142 source gateway 140 specifications H 323 gateway 143 speech packets silence compression 135 speech path setup 143 status H 323 terminals list 71 SWCA group answering 89 switches network prerequisites 29 Symbol see NetVision Symbols 13 system configuration Business Communications Manager prerequisites 34 System wide Call Appearance see SWCA T target lines VoIP trunks incoming calls 89 TCP traffic behavior 160 TDM see PSTN fallback enabled template file H 323 terminals list 71 terminal status 53 text conventions 14 time exceeded 141 TimeToLive 107 tips 13 Traceroute 141 156 traffic network loading 128 network mix 160 WAN link resources 121 transfer media path redirection 144 transmission characteristics 122 transmit fallback threshold 154 transmit path 124 Transmit Threshold 78 88 92 troubleshooting IP telephones 48 network delay and error analysis 131 Sniffer 142 trunks VoIP 20 two way call bandwidth requirements 123 IP Telephony Configuration Guide 180 In
80. To reset the Hot Desking password field for a specific IP telephone 1 Click on the key beside Services and IP Telephony 2 Click on Nortel IP Terminals 3 Click on the IP Terminal Status tab Figure 11 IP Terminal Status tab list Edit Configuration Performance Fault Report Tools Logoff View Help IP Terminal Status IP Terminal Status 2431 Offline i2050 N Default IN Default IN 2432 Offline i2004 N A Default N A Default INA 2433 Ofline 2002 N A Defaut N A Default INA IP Telephony Configuration Guide 58 Chapter3_ Installing IP telephones 4 Select the IP telephone record you want to reset 5 On the top menu click Configuration then select Reset Hot Desking Password Performance Fault Modify parameters Deregister DN F fi Add Feature Modify Feature Delete Feature 6 Adialog box appears prompting you to proceed Click Yes to reset the password The password resets to Null The user can enter hot desking again to enter a new password Notes about Hot Desking The Hot Desking feature allows a user to divert calls and signals from one IP telephone to another For instance if a user is temporarily working in another office they can retain their telephone number by hot desking their usual telephone to the IP telephone in their temporary office Hot desking can be accessed using FEATURE 999 on the telephone to which the traffic will be diverted The user can also evok
81. VLAN IDs to the IP telephones enter VLAN A none the entry must be terminated with a period e Summary tab Status box is set to Enabled 2 Ensure that the DHCP LAN settings are correct DHCP Local Scope LANX where LANX is a LAN that contains IP sets that use DHCP e Scope Specific Options tab Scope Status Enabled Default Gateway Field Published IP Address Address Range tab contains the range of IP addresses you need 3 Restart all existing connected IP telephones Note Whenever changes are made to the DHCP settings telephones will retain the old settings until they are restarted If the DHCP server is not properly configured with the Published IP address the telephones will display Invalid Server Address lfthis message appears correct the DHCP settings and restart the telephones IP telephony DHCP notes The 12004 supports two forms of DHCP configuration full and partial If partial DHCP is selected the user must manually enter the primary and secondary Business Communications Manager address action retry count The 12004 then configures a IP address netmask and default IP gateway via DHCP If full DHCP is selected the 12004 configures all parameters via DHCP Note If partial DHCP is selected the DHCP server does not need to send the vendor specific or site specific information outlined below The information below pertains to Full DHCP only In the case of partial DHCP the 12004 requires only the Rou
82. a bay modules on the Business Communications Manager 3 0 e Configuration of lines is complete e Operators have a working knowledge of the Windows operating system and of graphical user interfaces e Operators who manage the data portion of the system are familiar with network management and applications Refer to Chapter 2 Prerequisites checklist on page 29 for more information Symbols used in this guide This guide uses these symbols to draw your attention to important information Caution Caution Symbol Alerts you to conditions where you can damage the equipment Danger Electrical Shock Hazard Symbol AN Alerts you to conditions where you can get an electrical shock Warning Warning Symbol N Alerts you to conditions where you can cause the system to fail or work improperly IP Telephony Configuration Guide 14 Preface Note Note Tip symbol Alerts you to important information Tip Note Tip symbol Alerts you to additional information that can help you perform a task Text conventions This guide uses these following text conventions angle brackets lt gt Represent the text you enter based on the description inside the brackets Do not type the brackets when entering the command Example If the command syntax is ping ip address youenter ping 192 32 10 12 bold Courier text Represent command names options and text that you need to enter Example Use
83. a full duplex link with silence compression 140 Relationship between users and services 02 000s eee ee eee 154 P0993474 03 11 Tables Table 1 Network diagram prerequisites 0 cee eee 29 Table 2 Network device checklist cccccter cate ee RR esate teen EEG ORE tans 30 Table 3 Neiwork assessment ois uae ke edeen seek ot dug doge Sox de qeu dos dede we 30 Table 4 ligsuulco ASSESSING lusdeqenxaqadccsRRRressshbeewdugssdea dq iq 31 Table 5 IP terminals general record fields 0 000 eee eens 32 Table 6 ur TCR 23 Table 7 Business Communications Manager system configuration 34 Table 8 IF telephone provisioning ssec terme ER RE RDURRRE RR REEES 37 Table 9 IP terminals general record fields 0 000 e eee eee eee 41 Table 10 IP telephone server configurations llle 46 Table 11 IP telephony display messages seeeselln ene 48 Table 12 IP telephone troubleshooting cece eee eee 48 Table 19 IP Terminal Status fields cao ccc eeeeae RR ERREUR eddaes caves nas 54 Table Te Relabelling examples uude woacdot uda cenba X day Qin bl ode Rie dd awe eek 60 Table 15 H 323 Terminal lal csc ieee aces vada Rx RR RR dee NE ERNE ERE Eee ee 70 Table 16 Media parameters record 000 0c eee tee eee 75 Table 17 Remote gateway record 0 cece ee eens 79 Table TS QOS SIGS class caatc acne Eanes RR br y Eph ERRAR EUR Ra d
84. able 37 Software network communications application compatibility 152 Table 38 Quality of voice Service auus pack nae CR ORo tnam eq OES A 154 Table 398 Ste pairs and routes co cc ccs cceeree es ea IRR caw es Xe DU X Cao ks 156 Table 40 Computed load of voice traffic per link llle 157 Table 41 Delay and gor SIdIBIICe esesseueoudto see Ee c RR OPEN Ue ED QU cade 158 IP Telephony Configuration Guide 12 Tables P0993474 03 13 Preface This guide describes IP Telephony functionality for the Business Communications Manager 3 0 system This includes information about Nortel IP terminals such as the 12002 12004 telephone and the Nortel Networks 12050 Software Phone the Symbol NetVision and NetVision data telephones H 323 protocol devices and VoIP trunks and H 323 trunking with such applications as NetMeeting Before you begin This guide is intended for installers and managers of a Business Communications Manager 3 0 system Prior knowledge of IP networks is required Before using this guide the Business Communications Manager 3 0 system must be configured and tested This guide assumes e You have planned the telephony and data requirements for your Business Communications Manager 3 0 system e The Business Communications Manager 3 0 is installed and initialized and the hardware is working External lines and internal telephones and telephony equipment are connected to the appropriate medi
85. accesses an Ottawa outside line IP Telephony Configuration Guide 96 Chapter5 Configuring VoIP trunks Connecting an i200X telephone This section takes the example above and uses it to demonstrate how an installer would configure an 12002 or i2004 telephone on the system For information on configuring i200X telephones see Chapter 3 Installing IP telephones on page 39 Note IP clients require an IP network to reach the Business Communications Manager However they do not need to use VoIP trunks to communicate beyond the Business Communications Manager They can use any type of trunk just as any other phone on the Business Communications Manager can Connecting an i200X telephone on the LAN In this case the Santa Clara administrator wants to connect an 12004 phone using the LAN 1 network interface 1 The installer sets up the Business Communications Manager to handle the IP telephone by turning Registration to ON and Auto Assign DNs to ON 2 The installer connects the telephone to the LAN and sets it up using the following settings e Set IP address 10 10 5 10 e Default GW 10 10 5 1 This is the IP address of the default gateway on the network which is the nearest router to the telephone e S1 IP address 47 62 84 1 This is the published IP address of the Business Communications Manager The Business Communications Manager automatically assigns the telephone the DN of 3348 The installer configures DN r
86. aces field Select a template in the Templates field Type a descriptive name in the Filter Name field Click OK The Priority Outbound Filters window opens Click Apply The filter is applied to the circuit Sample criteria ranges and actions for UDP filtering The filtering goal is to place all VoIP H 323 traffic leaving a particular interface in the high priority queue From the BayRS Site Manager Use a criteria path of Criteria Add IP IP UDP Destination Port The range is 28000 to 28255 The action path is Action IP Add High Queue Note This example shows how to give H 323 traffic priority over other protocols on the interface IP Telephony Configuration Guide 148 Interoperability Using VLAN on the network A virtual LAN VLAN is a logical grouping of ports controlled by a switch and end stations such as IP telephones configured so that all ports and end stations in the VLAN appear to be on the same physical or extended LAN segment even though they may be geographically separated VLAN IDs are determined by how the VLAN switch is configured If you are not the network administrator you will have to ask whoever manages the switch what the VLAN ID range is for your system VLANs aim to offer the following benefits e VLANs are supported over all IEEE 802 LAN MAC protocols and over shared media LANs as well as point to point LANs e VLANs facilitate easy administration of logical groups of s
87. age 71 e Changing a handset Name on page 72 e Changing the DN record of a handset on page 72 e Deleting a NetVision telephone from the system on page 72 NetVision connectivity The Business Communications Manager supports access points NetVision handsets and other wireless IP devices that use either IEEE 802 11 1 or 2 M bits sec Frequency Hopping Spread Spectrum or IEEE 802 11B 11 M bits sec Direct Sequence Spread Spectrum technology NetVision telephones use an enhanced version of H 323 referred to as H 323 NetVision and NetVision Data wireless IP telephones connect to the Business Communications Manager over a LAN through the Business Communications Manager LAN or WAN card The Business Communication Manager sees these telephones as IP telephones which means that the DN records are assigned from the digital range rather than from the Companion or ISDN range of DNs The default codec for NetVision handsets is G 729 However if the NetVision handsets connect over IP trunks the codec of the IP trunk takes precedence Note NetVision handsets experience communications problems if your system has a NAT between the handset internet connection and the published address of the Business Communications Manager LAN For this reason this configuration is NOT supported From within the system the handsets can make and receive calls from any trunk type supported by the system which can include voice over IP VoIP digit
88. ager S2 IP ip address This is the Published IP address of the second Business Communications Manager that you want to register the telephone to It can also be the same as the S1 setting S2 PORT Default 7000 This is the port the telephone will use to access this Business Communications Manager S2 ACTION Default 1 S2 RETRY COUNT digits between 0 and 255 Set this to the number of times you want the telephone to retry the connection to the Business Communications Manager P0993474 03 Chapter 3 Installing IP telephones 47 Table 10 IP telephone server configurations Continued Field Value Description VLAN 0 No VLAN Choose 0 NO VLAN if there is no VLAN on the network 1 Manual VLAN 2 Automatically discover VLAN using DHCP If you do not have DHCP on the network or if DHCP is supplied by a remote server select number 1 and enter the VLAN ID If you have the Business Communications Manager DHCP active on your system select number 2 if you want DHCP to automatically find the VLAN assignment Refer to Configuring DHCP on page 49 VLAN is a network routing feature provided by specific types of switches To find out if VLAN has been deployed on your system check with your network administrator If VLAN is deployed the system administrator responsible for the switch can provide the VLAN ID s for your system Refer to the Programming Operations
89. al and analog trunks The handset DN record determines which lines the handset can access The handset can communicate with any other type of telephone supported by the Business Communications Manager system IP Telephony Configuration Guide 68 Chapter4 Installing NetVision telephones Access points Instructions about installing an 802 11b access point are provided with the access point equipment which is sold and installed separately The access point is set up with a unique identifier ESS ID which is entered into the handset either through a configuration download or manually through the dialpad to allow the handset to access the system through that access point Keycodes Before setting up NetVision telephones ensure that you have enough IP client keycodes enabled to register all the NetVision telephones you require For information about entering keycodes see the Keycode Installation Guide IP clients are distributed on a one to one basis with NetVision and IP telephones so ensure that you take your entire system into consideration Handset and call functions Symbol supplies a handset user guide that describes the features on the NetVision handset and how to use them to perform basic functions The Business Communications Manager NetVision Feature card explains how to use the handset to access features on the Business Communications Manager system and provides some quick tips for basic call functions The Business Comm
90. andwidth in kbit s IP Telephony Configuration Guide 132 Efficient Networking Queuing delay The queuing delay is the time it takes for a packet to wait in the transmission queue of the link before it is serialized On a link where packets are processed in a first come first served order the average queuing time is in milliseconds and is the result of the following formula queuing time in ms 8 average IP packet size in bytes 1 p link bandwidth in kbit s The average size of intranet packets carried over WAN links generally is between 250 and 500 bytes Queueing delays can be important for links with bandwidth under 512 kbit s while with higher speed links they can allow higher utilization levels Routing and hop count Each site pair takes different routes over the intranet The route taken determines the number and type of delay components that add to end to end delay Sound routing in the network depends on correct network design Reduce link delay In this and the next few sections the guidelines examine different ways of reducing one way delay and packet loss in the network The time taken for a voice packet to queue on the transmission buffer of a link until it is received at the next hop router is referred to as the link delay Methods to reduce link delays include Upgrade link capacity to reduce the serialization delay of the packet This also reduces the utilization of the link reducing the queueing delay Bef
91. anet route Santa Clara Richardson R1 R4 R5 R6 Santa Clara Ottawa R1 R2 Santa Clara Tokyo R1 R4 R5 R7 Richardson Ottawa R2 R3 R5 R6 P0993474 03 Quality of Service 157 Table 40 Computed load of voice traffic per link Links Traffic from R1 R4 Santa Clara Richardson Santa Clara Tokyo R4 R5 Santa Clara Richardson Santa Clara Tokyo R5 R6 Santa Clara Richardson Richardson Ottawa R1 R2 Santa Clara Ottawa R5 R7 Santa Clara Tokyo R2 R3 Richardson Ottawa R3 R5 Richardson Ottawa Adjusting Ping measurements The Ping statistics are based on round trip measurements While the QoS metrics in the Transmission Rating model are one way To make the comparison compatible the delay and packet error Ping statistics are halved Adjustment for processing The Ping measurements are taken from Ping host to Ping host The Transmission Rating QoS metrics are from end user to end user and include components outside the intranet The Ping statistics for delay requires additional adjustments by adding 140 ms to explain the processing and jitter buffer delay of the gateways No adjustments are required for error rates If the intranet measurement barely meets the round trip QoS objectives the one way QoS is not met in one of the directions of flow This state can be true when the flow is on a symmetric route caused by the asymmetric behavior of the data processing services Late packets Packets that arrive outsid
92. as enough information to load the IP telephony traffic on the intranet Consider the intranet has the topology as shown in the figure below and the installer or administrator wants to know in advance the amount of traffic on a specific link RA R5 Figure 44 Calculating network load with IP telephony traffic L Santa Clara Santa Clara Richardson traffic 4 Ottawa Tokyo traffic Santa Clara Tokyo traffic Richardson Business Communications Manager IP telephony 12 Router P0993474 03 Efficient Networking 129 Each site supports four VoIP ports Assume the codex is G 729 Annex B 20 ms payload Assuming full duplex links peak bandwidths per call are between 24 8 kbit s and 27 6 kbit s peak transmission or approximately 28 kbit s This is shown in the following figure taken from the table under Bandwidth requirements on full duplex links on page 125 Figure 45 Network loading bandwidth PPP B W FR B W Payload Silence Silence Size NoSP Suppression No SP Suppression Codec ms peak peak Avg peak peak Avg Type kbit s kbit s kbit s kbit s kbit s kbit s G 729 10 41 6 20 8 47 2 23 6 8 kb s 20 24 8 12 4 27 6 13 8 30 19 2 9 6 21 1 10 6 Route R1 R2 needs to support four VoIP Calls R4 R5 needs to support eight VoIP calls The incremental peak bandwidth for VoIP traffic is therefore R1 R2 peak VoIP Load 4 28 kbit s 112kbit s R4 R5
93. ass Outgoing and Block Incoming Except IP Phones this only allows IP telephony registration traffic through but blocks all other traffic including H 323 calls on this interface You must still specify an H 323 rule to allow IP call voice traffic Also Registration must be turned on in the Services IP Telephony Nortel IP Telephone General page before the telphone can access the System to register Change the contrast level When an IP telephone is connected for the first time the contrast level is set to the default setting of 1 Most users find this value is too low Therefore after the telephone is installed use FEATURE 9 and use the UP or DOWN key to adjust the contrast Block individual IP sets from dialing outside the system If you want to block one or more IP telephones from calling outside the system use Restriction filters and assign them to the telephones you want to block Restriction filters are set up under Services Telephony Services Restriction filters Restriction filters are discussed in the Programming Operations Guide Configuring DHCP You can use DHCP to automatically assign IP addresses to the IP telephones as an alternative to manually configuring IP addresses for IP telephones If you are using the Business Communications Manager as the DHCP server you can also configure the server to automatically locate the VLAN ID for the system and assign it to the telephones that register Befor
94. ation Guide 82 Chapter5 Configuring VoIP trunks Setting up the VoIP schedule You can determine which telephones lines will choose the VoIP route as the prime route by setting up the VoIP schedule to allow you to manually activate the service from a control set The PSTN route gets assigned to the Normal schedule which runs on all telephones when no other schedule is activated Rename Schedule 4 Services Telephony Services Scheduled Services Common Settings Schedule Names to VoIP Refer to the Programming Operations Guide for detailed instructions if required Then follow these steps to set up the VoIP schedule for routing services 1 Open Services Telephony Services Scheduled Services Routing Service and click on VoIP The VoIP schedule screen appears in the right frame Figure 22 VoIP Routing Service i2 Lo ive VoIP Del All r VoIP P Routing service Night Service setting Manual x Evening i Lunch Overflow yi M VoIP Sched 5 2 Change the Service setting to Manual 3 Change the Overflow setting to Y Configuring routes Configuring routes allows you to set up access to the VoIP and the PSTN line pools These routes can be assigned to destination codes using schedules Note If you have not already done so remember to define a route for the local PSTN for your own system so users can still dial local PSTN numbers Ensure the PSTN and VoIP line pools have been configured bef
95. ations Manager 3 0 currently initiates calls using H 323 fastStart methods The Business Communications Manager will accept and set up calls that have been initiated by another endpoint using H 323v2 fastStart methods as well as H 323 slowStart methods Media path redirection Media path redirection occurs after a call has been established when an attempt is made to transfer to or conference in another telephone Business Communications Manager 3 0 does not support codec re negotiation upon media path redirection To ensure that call transfers and conference works correctly the following rules must be followed e The first preferred codec for VoIP Trunks must be the same on all Business Communications Managers See Configuring media parameters on page 74 If this codec is G 729 or G 723 the Silence Suppression option must be the same on all Business Communications Managers involved P0993474 03 Interoperability 145 e finterworking with a Meridian 1 ITG the profile on the Internet Telephony Gateway ITG must be set to have the same first preferred codec as on the Business Communications Manager the Voice Activity Detection VAD option must be set to the same value as the Silence Suppression on the Business Communications Manager and the ITG payload size must be set to 30 ms If these rules are not adhered to simple calls will still go through but some transfer scenarios will fail Gatekeeper The Business Commun
96. ave allocated sufficient resources on the Business Communications Manager for IP telephony For information about changing the DS30 split for the Business Communications Manager and allocating media resources refer to the Business Communications Manager Programming Operations Guide data sections Table 4 Resource assessment Prerequisites Yes No 4 a Has a Business Communications Manager Resource Assessment been performed using the resource questionnaire in the Programming Operations Guide 4 b Has an analysis been done to determine which DS 30 split is appropriate for the system Has the DS30 split been changed to 3 5 if necessary 4 c Have all necessary media resources for IP trunks clients vmail or WAN dialup been assigned or dedicated 4 d Have the necessary media gateway IP client and IP trunks resources been set Refer to Configuring media gateway parameters for IP service IP Telephony Configuration Guide 32 Chapter2 Prerequisites checklist Configuring media gateway parameters for IP service To set up the media gateway resources that you require for optimum IP telephony set the fields on the System Configuration window 1 Click the keys beside Services IP Telephony 2 Click on System Configuration The Parameters screen appears in the right frame Figure 2 System Configuration Parameters screen aAa 3 m 3 iensive Parameters ete Del All Para
97. ayload of 20 ms Silence compression is enabled The Ethernet LAN is half duplex Ethernet LAN may also be full duplex Given the above what is the peak traffic in kbit s that IP telephony will put on the LAN From the table under Bandwidth requirements on half duplex links on page 124 the following figure shows the peak transmission bandwidth for G 729 with silence suppression enabled on a half duplex link is 34 4 kbit s per call or 137 6 kbit s for all four calls Figure 42 LAN engineering peak transmission Ethernet B W Silence NoSP Suppression peak peak Avg kbit s kbit s kbit s G 729 10 8 kb s 20 34 4 34 4 30 WAN engineering Wide Area Network WAN links are typically full duplex links both talk and listen traffic use separate channels For example a T1 link uses a number of 64 kbit s DSO duplex channels allowing 64 kbit s for transmit path and n 64 kbit s for the receive path WAN links may also be half duplex Example 1 WAN engineering voice calls Consider a site with four IP telephony ports and a full duplex WAN link using PPP The preferred codec is G 729 kbit s which uses a voice payload of 20 ms Silence compression is enabled Given the above what is the peak traffic in kbit s that IP telephony will put on the WAN From the table under Bandwidth requirements on full duplex links on page 125 the following figure shows the peak transmissio
98. been set up to allow Refer to IP telephones on page IP telephones 37 If you are configuring the Business Communications Manager DN records records before you configure the handset You know which DNs you want to assign to the handsets and you have all the line restrictions and telephony information you require to create or update a DN record for each telephone Download the latest version of the NetVision Phone Administrator http www symbol com services downloads nvfirmware2 html Download the latest firmware version from the same website You have obtained the Symbol NetVision serial cable which is Purchased from Symbol at lt http used to transfer configuration information between the computer symbol com where the tool is installed and the handset part number 25 20528 01 You have a list of names that you will use for the handsets Each Name field name must be unique to a handset Both the H 323 Terminals record and the NVPA record must have exactly the same name You have identified a PIN for each handset Password field Assigning H 323 Terminals records The H 323 Terminals record Services IP Telephony identifies the NetVision handsets within the Business Communications Manager The Business Communications Manager uses the information from this file to determine if the handset will be allowed to connect to the system Notes The following are some notes about the proces
99. ca eae hc eek een eRe me e dese Re es 102 Figure 35 Local gateway IP interfaco oiseau scere io ee dee ik 105 Figure 36 Business Communications Manager systems with a gatekeeper 118 Figure 37 M1 to Business Communications Manager network diagram 116 Figure 38 Multiple Business Communications Manager systems network diagram 117 Figure 39 Routing all system wide public calls through one Business Communications Manager 118 Figure 40 M1 to Business Communications Manager network diagram 119 IP Telephony Configuration Guide 10 Figures Figure 41 Figure 42 Figure 43 Figure 44 Figure 45 Figure 46 Figure 47 Figure 48 Figure 49 Figure 50 Figure 51 Figure 52 Connecting to IP telephones 200 e eee eee ee 120 LAN engineering peak transmission 000 0c eee eee eee ees 126 Peak traffic WAN INK 2022660 32660485 exero ER Rx RER RE RE XE ERE 127 Calculating network load with IP telephony traffic llle 128 Network loading bandwidth ananuna ananena 129 One call on a half duplex link without silence compression 136 One call on a half duplex link with silence compression 136 Two calls on a half duplex link with silence compression 137 One call on a full duplex link without silence compression 138 One call on a full duplex link with silence compression 139 Two calls on
100. cally calculates the prevailing QoS level per site pair based on the measurement of the following e one way delay e packet loss e codec Table 38 Quality of voice service MOS Range Qualitative Scale 4 86 to 5 00 Excellent 3 00 to 4 85 Good 2 00 to 2 99 Fair 1 00 to 1 99 Poor P0993474 03 Quality of Service 155 When the QoS level of any remote gateway is below the fallback threshold all new calls are routed over the standard circuit switched network if fallback is enabled The computation is taken from the ITU T G 107 Transmission Rating Model Measuring Intranet QoS Measure the end to end delay and error characteristics of the current state of the intranet These measurements help to set accurate QoS needs when using the corporate intranet to carry voice services This section provides information about e Measuring end to end network delay on page 155 e Measuring end to end packet loss on page 156 Recording routes on page 156 e Adjusting Ping measurements on page 157 e Measurement procedure on page 158 e Other measurement considerations on page 158 Measuring end to end network delay The basic tool used in IP networks to get delay measurements is the Ping program Ping takes a delay sample by sending a series of packets to a specified IP address and then returning to the originating IP address Ping then displays statistics for the packets High packet times can i
101. ce The more hops that are required the more potential there is for voice quality issues to arise hot desking See Feature 999 hub Center of a star topology network or cabling system IEEE802 ESS This is the LAN and switch standard used to define the connection between the access point and the NetVision handset onto the network The handset is given the ID code of the device s with this standard so the access points recognize them i2050 Software Phone This is a computer based version of an IP telephone Once installed it acts and is programmed as you would the 12004 telephone You must have a sound card and a USB headset to use this application interoperability Interoperability refers to how compatible Business Communications Manager data configuration is with the rest of the network Business Communications Manager IP Telephony adheres to the ITU T H 323v2 standards and is compatible with any H 323v1 or H 323v2 endpoints This also refers to IP compatibility issues between released versions of the Business Communications Manager Business Communications Managers on the network with earlier versions of the software will not have the same operability for VoIP trunks as systems with 3 0 software IP Telephony Configuration Guide 168 Glossary IP server On the Business Communications Manager this is the server that registers IP telephones IP telephone In this book this term refers to any internet base
102. ces such as e address translation e call control e admission control ARQ IP Telephony Configuration Guide 104 Chapter5 Configuring VoIP trunks e bandwidth control zone management A single Gatekeeper manages a set of H 323 endpoints This unit is called a Gatekeeper Zone A zone is a logical relation that can unite components from different networks LANS These Gateway zones such as the Business Communications Manager are configured with one or more alias names that are registered with the gatekeeper The gatekeeper stores the alias IP mapping internally and uses them to provide aliases to IP address translation services Later if an endpoint IP address changes that endpoint must re register with the gatekeeper Refer to the gatekeeper software documentation for information about changing IP addresses Note The Business Communications Manager has been tested by Nortel Networks to be compliant with RADVISION ECS 2 1 0 1 http www radvision com and CSE 1000 gateway applications Note A gatekeeper may help to simplify IP configuration or the Business Communications Manager dialing plan however it will not simplify the network dialing plan Modifying the Local Gateway Settings The call signaling method used by the local gateway defines how the Business Communications Manager prefers call signaling information to be directed Call signaling establishes and disconnects a call If the network has
103. cklist for IP telephony 29 delay and error analysis 131 determining bandwidth 121 determining WAN link resources 121 efficiently 121 engineering link capacity 130 engineering worst case 122 IP address 36 LAN engineering examples 126 MCDN over VoIP 80 115 multi locations with call center 119 multiple Business Communications Manager 117 non linear processing 127 other internet resource considerations 131 PSTN fallback 80 remote IP telephone site 120 signaling method 104 transmission characteristics 122 using a gatekeeper 103 VoIP destination digits 80 WAN engineering 126 networks VLAN ports 37 NEW SET 48 no connection IP telephones 48 no speech path IP telephones 49 non linear processing 127 Nortel NVPA changing handset name 72 user name 69 NPI TON 106 number of calls usable link bandwidth 123 O one way delay 132 one way speech path IP telephones 49 outbound traffic filter creating 146 Outgoing call configuration 76 outgoing calls 76 overflow setting 82 P Packet delay 162 packet errors reducing 133 loss 122 132 162 queuing delay 132 Packet InterNet Groper see Ping password H 323 terminals list 71 hot desking 999 57 payload size 122 124 125 144 peak bandwidth 123 124 peak traffic 122 126 physical link capacity 122 Ping 141 155 157 planning modules 130 port settings 101 103 ports firewalls 101 legacy networks 103 PPP B W 122 124 125 preferred codec 75 pre installatio
104. corresponding IP clients P0993474 03 Chapter2 Prerequisites checklist 35 Defining published IP address The published IP address is the IP address used by computers on the public network to find the Business Communications Manager For example if a Business Communications Manager has a LAN interface LANI that is connected only to local office IP terminals and a WAN interface WAN1 that is connected to the public network then WANI should be set to the published IP address Setting the Global IP published IP To set the published IP address 1 InUnified Manager open Services and click on IP Telephony The Global settings tab appears as shown in the diagram below 2 From the Published IP Address menu select the appropriate network interface Figure 3 Global IP settings scel Global IP Setting lete Dei Al Global IP Setting ft 47 65 138 52 a Eim Published IP Address H323Address esources i ervices B Telephony Ser 9 IP Telephony System C Nortel IP Tia IP Telephony Configuration Guide 36 Chapter2 Prerequisites checklist Determining the published IP address Use the flowchart in the following figure to determine which card should be set as the published IP address Figure 4 Setting the Published IP address Set the network interface with Is NAT enabled N the most anticipated VoIP traffic as the Publishe
105. costs for upgrading the intranet The evaluation often requires a link upgrade a topology change or implementation of QoS in the network To maintain costs you can accept a Fair QoS level for the time for a selected route A calculated trade off in quality requires the installer or administrator to monitor the QoS level reset needs with the end users and respond to user feedback Implementing QoS in IP networks This section describes information about implementing QoS in IP networks e Traffic mix on page 160 e TCP traffic behavior on page 160 Business Communications Manager router QoS support on page 161 Corporate intranets are developed to support data services Accordingly normal intranets are designed to support a set of QoS objectives dictated by these data services When an intranet takes on a real time service users of that service set additional QoS objectives in the intranet Some of the targets can be less controlled compared with the targets set by current services while other targets are more controlled For intranets not exposed to real time services in the past but which now need to deliver IP telephony traffic QoS objectives for delay can set an additional design restriction on the intranet One method of determining requirements is to subject all intranet traffic to additional QoS restrictions and design the network to the strictest QoS objectives An exact plan for the design improves the quality
106. d IP address Y 2n Set the network interface on the Is the Business Communications N private side as the published Manager expected to connect to IP address devices on the public side Y Are all of your public side N Set the network interface on the devices using a VPN ee as the published Y Do you anticipate the most Public VOIP traffic on your public or private side Set the network interface on the public side as the published IP address Private Set the network interface on the private side as the published IP address The flowchart shown above makes reference to public and private IP addresses The public and private IP addresses are concepts relating to Network Address Translation NAT The decision also depends on whether a Virtual Private Network VPN is enabled For information about NAT and VPN refer to the Business Communications Manager 3 0 Programming Operations Guide If you use IP telephones on the network they must be set to have the IP address of the network card they are connected to for their Default Gateway and the Published IP address as the S1 IP address For more information about this see Configuring the 12002 or 12004 telephone to the system on page 44 P0993474 03 Chapter2 Prerequisites checklist 37 IP telephones Complete this section if you are installing I
107. d Manager firewalls Firewalls are server security devices on a network that block or allow IP traffic into node networks or devices When configuring IP telephony you need to ensure that the port settings are correctly configured to pass through any network firewalls between the telephone and the Business Communications Manager full duplex transmission Simultaneous two way separate transmission in both directions G 711 A codec that delivers toll quality audio at 64 kbit s This codec is best for speech because it has small delay and is very resilient to channel errors G 729 A codec that provides near toll quality at a low delay Uses compression to 8 kbit s 8 1 compression rate P0993474 03 Glossary 167 G 723 1 A codec that provides the greatest compression 5 3 kbit s or 6 3 kbit s Normally used for multimedia applications such as H 323 videoconferencing Allows connectivity to Microsoft based equipment gatekeeper A gatekeeper is server application on a network that tracks IP addresses of specified devices to provide authorization for making and accepting calls for those devices The Business Communications Manager supports RadVision and CSE 1000 gatekeeper applications H 323 The ITU standard for multimedia communications over an IP network Business Communications Manager IP Telephony supports H 323 hop count This is the number of routers the signal must go through to reach the destination devi
108. d at this step gt To move a Nortel IP telephone and change the DN 1 2 3 Deregister the DN using the instructions in Deregistering DNs for IP telephones on page 62 Disconnect the network connection and the power connection from the telephone Reinstall the phone at the new location and redconfigure the telephone For information about this see Connecting the 12002 or 12004 telephone on page 44 Keep DN alive This feature is only relevant to the i series IP telephones Model 12004 12002 12050 If you want to retain DN specific features such as Call Forward No answer and Call Forward on Busy if an IP telephone becomes disconnected you must ensure the following setting is set to Y in the Unified Manager 1 In the Unified Manager under the Services Telephony Services list click on the DN record for the IP telephone Click the Capabilities heading Beside the Keep DN alive field choose Y Choosing N for this field allows the DN record to become inactive if the IP telephone is disconnected This produces a Not in Service prompt if any of the special features such as Call Forward are invoked i Warning If the system is reset while an IP telephone is disconnected the Keep DN alive feature becomes inactive until the telephone is reconnected Note When an IP telephone is disconnected there is about a 40 second delay before the gt system activates Keep DN alive during which incoming calls
109. d receiver alternate the use of the shared channel the peak bandwidth requirement is equal to the full transmission rate Only one media path is present on the channel at one time Figure 47 One call on a half duplex link with silence compression Tx Hello Fred This is Susan Do you have a minute Fred here Hi Sure Rx Conversation d 5 Bandwidth sed Channel Link max d c Z yr N Z V X J Time in di Y Half duplex call alternates use of half duplex bandwidth P0993474 03 Silence compression 137 The effect of silence compression on half duplex links is therefore to reduce the peak and average bandwidth requirements by approximately 5046 of the full transmission rate Because the sender and receiver are sharing the same bandwidth this affect can be aggregated for a number of calls The following figure shows the peak bandwidth requirements for two calls on a half duplex link with silence compression enabled The peak bandwidth for all calls is equal to the sum of the peak bandwidth for each individual call In this case that is twice the full transmission rate for the two calls Figure 48 Two calls on a half duplex link with silence compression Tx Rx Tx Rx Tx Rx Chan Bandwidth Conversation Buenos noches Juan Muy bien y tu Hola Isabella Com o estas Hello Fred This is Susan Do you have a minute meaner Wm Sure Conversation 2 Bandwi
110. d telephone that works with the Business Communications Manager system For this release this includes the Nortel Networks IP telephones 12002 12004 and 12050 Software Phone as well as the Symbol NetVision sets and NetVision data handsets These telephones all interface to the Business Communications Manager LAN or WAN card through an internet or intranet link ITG This is the internet telephony gateway protocol for the Meridian 1 to Business Communications Manager IP trunk connections VoIP trunks require compatible configuration at both endpoints The Business Communications Manager must be set to recognize that the other end of the trunk is an MI ITG system jitter buffer This is the process of collecting and organizing data frames at the receiving end to provide balanced voice quality kbit s kilobits per second Thousands of bits per second keycodes These are software codes that release feature applications on the Business Communications Manager such as VoIP trunks IP telephony ports and MCDN latency The amount of time it takes for a discrete event to occur Mbit s Megabits per second Millions of bits per second MCDN This is a specific network protocol used on private networks between Business Communications Manager systems or between Business Communications Manager systems and Meridian systems The protocol only works on PRI SL 1 lines and on VoIP trunks The protocol is activated with a keycode modem
111. detailed description of the configurations required for tandeming a system over PRI lines Except for the VoIP trunk requirements the system and routing configurations would be similar P0993474 03 Chapter6 Typical network applications using MCDN 119 Multi location chain with call center In the installation shown in the following diagram one Business Communications Manager runs a Call Center and passes calls to the appropriate branch offices each of which use a Business Communications Manager A typical use of this would be a 1 800 number that users world wide can call who are then directed to the remote office best able to handle their needs Figure 40 M1 to Business Communications Manager network diagram Call Center fallback route Intranet VoIP trunk Branch Offices i2004 telephone To set up this system 1 Ensure that the existing network can support the additional VoIP traffic Coordinate a Private dialing plan between the systems On each Business Communications Manager 3 0 system e Set up outgoing call configuration for the VoIP gateway e Set up a remote gateway for other Business Communications Managers e Set phones to receive incoming calls through target lines e Configure the PSTN fallback and enable QoS on both systems IP Telephony Configuration Guide 120 Chapter6 Typical network applications using MCDN 4 Reboot each system 5 Set up a Call Center on t
112. dex U UDP port 141 portranges 103 private access code 86 private network MCDN 116 Unified Manager deleting handset record 72 destination codes 84 DN record 78 H 323 Terminals record 70 H 323 Trunks record 74 78 setting up target lines 89 trunk line data line pools 77 Unified Messaging 117 Universal Dialing Plan see UDP usable link bandwidth number of calls 123 V VAD silence suppression 135 VLAN 46 IP telephone 47 148 i series telephones 37 site specific options 149 Voice Activity Detection VAD 135 144 voice compression 143 voice jitter buffer 76 voice path silence suppression 124 voice quality codec 42 jitter buffer 43 VoIP DISA 73 gateway progress tones 146 gateway prerequisites 29 implementing QoS into network 159 load 129 MCDN network 80 schedule activating 87 schedule network example 93 schedule setting up 82 trunks configuring 73 VoIP trunks activating QoS monitor 87 activating VoIP schedule 87 adding to DN records 78 configuration 73 configuring incoming calls 89 configuring NetMeeting clients 98 connecting IP telephones 96 defaultlines 76 defined 20 destination codes 84 destination digits 80 example configuration 91 global IP 35 incoming call configuration 89 jitter buffers 76 keycodes 74 line pool 76 making calls 95 media parameters 74 networking IP address 36 networking multiple systems 117 networking remote IP telephone site 120 Outgoing call configuration
113. dth used nU Channel Link max T E E Time M J S J i4 d Peak channel bandwidth is n average Bandwidth shared by half duplex calls bandwidth per call IP Telephony Configuration Guide 138 Silence compression Silence compression on Full Duplex Links On full duplex links the transmit path and the receive path are separate channels with bandwidths usually quoted in terms of individual channels The following figure shows the peak bandwidth requirements for one call on a full duplex link without silence compression Voice packets are transmitted even when a speaker is silent Therefore the peak bandwidth and the average bandwidth used equals the full transmission rate for both the transmit and the receive channel Figure 49 One call on a full duplex link without silence compression Tx Rx channel Tx channel Bandwidth Bandwidth Hello Fred This is Susan Do you have a minute Fred here Hi Sure Conversation Pd Bandwidth used E Channel Link max ime Channel Link max Voice frames sent even when speaker is silent P0993474 03 Silence compression 139 When silence compression is enabled voice packets are only sent when a speaker is talking When a voice is being transmitted it uses the full rate transmission rate Since the sender and receiver do not share the same channel the peak bandwidth requirement per
114. dule the installer assigns the route created for the local PSTN access with absorb digits set to All e Under the VoIP schedule the installer assigns the route created for the local PSTN access with absorb digits set to All The installer dials FEATURE 873 and selects the VoIP schedule VoIP is now activated At this point the system is configured to make outgoing calls but it is not set up to receive incoming calls If there are no target lines set up the installer creates target lines for each telephone record or Hunt group Making calls From a set on Business Communications Manager Ottawa a caller dialing a set on Business Communications Manager Santa Clara must dial the destination code which includes the destination digits for the Business Communications Manager Santa Clara remote gateway and the DN of the set For example dialing 33322 would connect as follows 3is the destination code If a suitable level of QoS is available the call is routed through the VoIP trunks and through the remote gateway with destination digits of 3 The call is sent across the PDN using the IP address of the Santa Clara Business Communications Manager 3322 is linked to the target line associated with DN 3322 The call arrives at the phone with the DN 3322 If a user in Santa Clara wanted to make a local call in Ottawa they would dial 29 followed by the local Ottawa number The digit 2 accesses the remote gateway for the VoIP line The digit 9
115. e of the window allowed by the jitter buffer are discarded To determine which Ping samples to ignore calculate the average one way delay based on all the samples Add 300 ms to that amount This amount is the maximum delay All samples that exceed this one way delay maximum are considered late and are removed from the sample Calculate the percentage of late packets and add that percentage to the packet loss statistics IP Telephony Configuration Guide 158 Quality of Service Measurement procedure The following procedure is an example of how to get delay and error statistics for a specific site pair during peak hours Program a script to run the Ping program during the intranet peak hours repeatedly sending a series of 50 Ping requests Each Ping request generates a summary of packet loss with a granularity of 2 and for each successful probe that made its round trip that many rtt samples For a strong network there must be at least 3000 delay samples and 60 packet loss samples Store the raw output of the Ping results in a file Determine the average and standard deviation of one way delay and packet loss Repeat this for each site pair At the end of the measurements the results are as shown in the following table Table 41 Delay and error statistics Measured one way delay Measured packet loss ms 96 Expected QoS level Destination pair Mean Mean o Mean Mean o Mean Mean o Santa Clara 171 179 2 2
116. e other end hears the voice Installing i series telephones The Nortel Networks i series telephones can be configured to the network by the end user or by the administrator If the end user is configuring the telephone the administrator must provide the user with the required parameters A maximum of 90 IP telephones including Nortel Networks 12050 Software Phones and H 323 devices such as NetVision handsets can be connected on the Business Communications Manager system Before installing Before installing the 12002 or 12004 telephone ensure that ensure the telephone has the appropriate power supply for your region e if powered locally ensure the installation site has a nearby power outlet otherwise it can be powered through a Power Inline Patch Panel PiPP e the installation site has a 10 100 BaseT Ethernet connection e if you are using an IP telephone that does not have a 3 port switch ensure you have 10 100 BaseT Ethernet connections for both the telephone and for your computer equipment Caution Do not plug the telephone into an ISDN connection This can cause severe damage to the telephone Plug the telephone only into a 10 100 BaseT Ethernet connection Using a 3 port switch In an office environment where a LAN network already exists most computers will already be connected to a LAN line To avoid the necessity of installing duplicate network connections you can use a Nortel Networks 3 port switch for o
117. e ports specified for VoIP are allowed A Nortel Networks 12002 or 12004 telephone uses ports between 51000 and 51200 to communicate with the Business Communications Manager The Business Communications Manager by default uses ports 28000 to 28255 to transmit VoIP packets IP Telephony Configuration Guide 102 Chapter5 Configuring VoIP trunks Follow these steps to modify these settings 1 In Unified Manager open Services IP Telephony Port Ranges The Port Ranges screen appears Figure 33 Port Ranges 3513852 a phony Services 2lephony 3ystem Configu Nortel IP Termin 4 323 Terminals 4 323 Trunks oriRanges 2 Select the Port Range you want to modify 3 From the top menu click Configuration and then select Modify PortRanges The Modify PortRanges dialog box appears Refer to Figure 34 Figure 34 Port ranges dialog box pice m NET i P0993474 03 Chapter 5 Configuring VoIP trunks 103 4 Change the port settings 5 Click the Save button Port settings for legacy networks Business Communications Manager 3 0 uses UDP port ranges to provide high priority to VoIP packets in existing legacy IP networks You must reserve these same port ranges and set them to high priority on all routers that an administrator expects to have QoS support You do not need to reserve port ranges on DiffServ networks You can select any port ranges that are not used by well known protocols or applicati
118. e setting up DHCP using the information below refer to the Business Communications Manager 3 0 Programming Operations Guide for detailed information about DHCP Note Do not enable DHCP on the Business Communications Manager if you have another DHCP server on the network Refer to the Business Communications Manager 3 0 Programming Operations Guide for detailed information about disabling DHCP or about using other types of DHCP To set up DHCP to work with IP terminals refer also to IP telephony DHCP notes on page 50 1 Ensure that DHCP under Services is set up with the following settings e Global Options tab NORTEL IP Terminal Information box is set to Nortel i2004 A ip address 7000 1 250 ip address 7000 1 1 Where ip address is the published IP address Be sure to include the period at the end of the string 1 250 IP Telephony Configuration Guide 50 Chapter3 Installing IP telephones Nortel IP Terminal VLAN ID contains an identification if the system is using the VLAN option If you do not know what the entry should be contact the system administrator for the VLAN switch If you want DHCP to automatically assign VLAN IDs to the IP telephones enter the VLAN IDs in the following format VLAN A id1 id2 idn Example if your VLAN IDS are 1100 1200 1300 and 1400 enter VLAN A 1100 1200 1300 1400 the entry must be terminated with a period If you do not want DHCP to automatically assign
119. e system can provide moderate quality e Receive Threshold 3 0 This is MOS value that ensures that the VoIP lines are used as long as the system can provide moderate quality e Destination Digits 2 Note In this case because the systems are on a CDP network and the 2 is included in the gt DN this number will be absorbed before dialout 7 The installer sets up the VoIP schedule with these settings e Service Manual Overflow Y P0993474 03 Chapter 5 Configuring VoIP trunks 95 10 11 12 The installer sets up the routes if they do not already exist The installer ensures a route has been created to the line pool that accesses the local PSTN line including the external dialout The installer defines a new route called Route 003 and sets it to use PRI A This is the line pool that contains the PRI fallback lines The installer defines a new route called Route 100 and sets it to use line pool O This is the line pool that contains the VoIP lines The installer creates a destination code of 2 e Under the Normal schedule the installer assigns Route 003 which uses line pool PRI A The absorb digits is set to All e Under the VoIP schedule the installer assigns Route 100 which uses the VoIP lines in line pool O The absorb digits is set to 0 The installer creates a destination code of 9 which is the line pool access code for the local PSTN access lines e Under the Normal sche
120. e this feature from the Services key menu where it is defaulted as the first item on the list Hot desking must be allowed on the originating telephone and you need to specify a password These settings are found under the ADMIN key within the hot desking feature Hot desking is invoked through the DIVERT key within the hot desking feature If the originating telephone does not have hot desking allowed the user will receive a Not allowed prompt indicating that the telephone is not available for hot desking This prompt also occurs if the originating telephone is on a call when the diversion command was issued Once hot desking occurs between two IP telephones no activity is allowed on the originating telephone except to cancel hot desking The display on the originating telephone indicates where it has been diverted On the diversion telephone the key displays will reflect the displays from the originating telephone Call forwarding to voice mail continues as normal Voice mail can be accessed from the active IP telephone as if it were the originating telephone When hot desking is cancelled which can be performed from either telephone the displays for each telephone return to normal If you forget the password hot desking can only be cancelled from the originating set Note When you cancel hot desking ensure that the telephone is on hook If you have just hung up wait 10 seconds before attempting to cancel hot desking
121. ecord 3348 with the lines and attributes the IP telephone requires The installer sets up a target line for DN 3348 using the Received Digits 3348 This phone would follow all of the same dialing rules as the other telephones on the Santa Clara Business Communications Manager A caller could dial 3321 to connect with telephone 3321 dial 9 to access the PSTN or dial 2 lt DN gt to access a telephone on the Ottawa system P0993474 03 Chapter 5 Configuring VoIP trunks 97 Example PSTN call to remote node Programming for tandeming Business Communications Managers together using PRI SL 1 lines and MCDN protocol is described in detail in the Programming Operations Guide Private Networking section VoIP tandem trunks are configured in the same way with the addition gateway programming required for IP trunks which is covered in Configuring a remote gateway on page 78 Making a call to a remote node requires that the receiving Business Communications Manager has the correct routing to pass the call on to the next node When the call is received system A the system recognizes that the received number is not a system number However if it has a route and destination code that recognizes the received number system A will then pass the call through Further the call is received as a public call on system A the call then becomes a private call as it is passed through a dedicated trunk in this case a VoIP trunk to system B On system B
122. ed RR y Rd ele RR 91 On Business Communications Manager Ottawa l i 92 On Business Communications Manager Santa Clara 00 cee eee 94 DAskilitreslli uses Sobre pv Vidi oo deii pe dub C odo RO Hdd eg qo eio 95 Connecting an 200 telephone uuu s eau e e ea rx RERO Rar RR C ER E RR Rn 96 IP Telephony Configuration Guide 6 Table of Contents Connecting an i200X telephone on the LAN 000 e eee eee 96 Example PSTN call ta remote ode j2ces opined eee dhe wi ee REG ES Ra RR danse RR URS 97 Remote access over VoIP fr nKS iizooss sue RR Epl e 9 xy eh oriri i43 97 Configuring Net Meeting clients 22 llus cus bs iwi eye ke bx Rc Rom Rc ed EORR 98 Quality of Service MONON uuusuessobuucscbG i kR ek X XXE ER we MEEKER ed dq 100 Quality of Service SIaliie iissoasat ede GR E RAGRcREAR KG e badd e REGERE EE eS 100 Updating the Dos monitor data cois x Lagar hod ole eee aote gg 100 Viewing QoS monitoring logging sss sees se Rx ex Rx RR eh xe 101 Fart SONNE vs x ud sedans Sek yee Bh SN See EIN aoa Epp eee UA UE e UNO ee ES 101 LPs seq da paa Pao TOR EEQUERDFEX E REQEIASPIG Hoes Gags aes Dame 101 Port settings for legacy networks 2 ciescevesdiwwrceaveotwrerewvoswus 103 Uenga ci o o pm Cmm 103 Modifying the Local Gateway Settings 0 eee ete elle 104 Notes about NPI TON alases iicisas sn e Rr y RR RR EE Eu RAS 107 Using Radvision ECS 2 1 0 1 as the gatekeeper lllsselss
123. ed by Business Communications Manager 3 0 for transmission Other values listed indicate payload sizes that Business Communications Manager can receive 2 Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter frame gap 3 G 711 does not support silence suppression Therefore the average bandwidth is the same as the peak bandwidth 4 Bandwidths stated per channel Rx or Tx With no silence suppression both the transmit path and the receive path continuously transmit voice packets Enabling silence suppression on full duplex links reduces the average bandwidth However since transmit and receive paths use separate channels the peak bandwidth per call per channel does not change Therefore peak bandwidth requirements per channel Rx or Tx per call on a full duplex link is Peak Bandwidth per channel per call 2 Continuous Transmission Rate Full Duplex links With or Without Silence Suppression The bandwidth made available by silence suppression on full duplex links with continuous transmission rate average bandwidth requirement is available for lower priority data applications that can tolerate increased delay and jitter IP Telephony Configuration Guide 126 Efficient Networking LAN engineering examples Example 1 LAN engineering voice calls Consider a site with four Business Communications Manager IP telephony ports Assume a preferred codec of G 729 which uses a voice p
124. ed physical lines Refer to Chapter 6 Typical network applications using MCDN on page 115 P0993474 03 Chapter 1 Introduction 23 In the figure above note that Business Communications Manager A is connected to a LAN through a LAN card to WAN through a WAN card and to a PSTN through trunk media bay modules Through these networks the system accesses other systems and network equipment connected to the network M1 ITG The Meridian 1 Internet Telephony Gateway M1 ITG allows Meridian 1 systems to communicate with the Business Communications Manager 3 0 via H 323 trunks In Figure 1 on page 22 telephones on the M1 such as Meridian telephone A can initiate and receive calls with the other telephones on the system across IP networks To provide fallback at times when IP traffic cannot pass you can also connect the Meridian to the Business Communications Managers through ISDN PRI SL 1 lines which provide the same MCDN capability that you can achieve through the VoIP trunks with MCDN active Refer to the Business Communications Manager Programming Operations Guide for a description of MCDN features and networking with PRI SL 1 lines Typical network applications using MCDN on page 115 describes how to provide the same network over VoIP lines A Business Communications Manager connected to an M1 ITG using the MCDN protocol can provide access to a central voice mail and call attendant systems which can streamline mul
125. eduction in average bandwidth requirements develops over a 20 to 30 second period as the conversation switches from one direction to another When a voice is being transmitted it uses the full rate or continuous transmission rate The effects of silence compression on peak bandwidth requirements differ depending on whether the link is half duplex or full duplex IP Telephony Configuration Guide 136 Silence compression Silence Compression on Half Duplex Links The following figure shows the bandwidth requirement for one call on a half duplex link without silence compression Since the sender and receiver share the same channel the peak bandwidth is double the full transmission rate Because voice packets are transmitted even when a speaker is silent the average bandwidth used is equal to the full transmission rate Figure 46 One call on a half duplex link without silence compression Tx Hello Fred This is Susan Do you have a minute Fred here Hi Sure Conversation x Bandwidth used Voice frames sent even when speaker is silent When silence compression is enabled voice packets are only sent when a speaker is talking In a typical voice conversation while one speaker is talking the other speaker is listening a half duplex conversation The following figure shows the peak bandwidth requirements for one call on a half duplex link with silence compression enabled Because the sender an
126. egrades the perceived voice quality To minimize this problem configure the IP telephone with a jitter buffer for arriving frames This is how the jitter buffer works Assume a jitter buffer setting of five frames e The IP telephone firmware places the first five arriving frames in the jitter buffer e When frame six arrives the IP telephone firmware places it in the buffer and sends frame one to the handset speaker e When frame seven arrives the IP telephone buffers it and sends frame two to the handset speaker The net effect of using a jitter buffer is that the arriving packets are delayed slightly in order to ensure a constant rate of arriving frames at the handset speaker P0993474 03 Chapter 1 Introduction 27 This delaying of packets can provide somewhat of a communications challenge as speech is delayed by the number of frames in the buffer For one sided conversations there are no issues However for two sided conversations where one party tries to interrupt the other speaking party it can be annoying In this second situation by the time the voice of the interrupter reaches the interruptee the interruptee has spoken 2 jitter size frames past the intended point of interruption In cases where very large jitter sizes are used some users revert to saying OVER when they wish the other party to speak Possible jitter buffer settings and corresponding voice packet latency delay for the Business Communicatio
127. em 000 eee eese 72 Chapter 5 Configuring VolP Irunks iiusuasesscukakouE cadens wt S4 REX ER AEZXSEA 73 Pre ingtallation system requirements cccc ccne ea sna dee kk bk ox RORCAC A A ORC aw ee 74 B CUOUR iae e y EN de d Dai d aw dae ae qd qq eq qe qid 74 Published IF AIOS ics up ek ERR FE ICE XAR RRARCOEX ACE Rd eevee dara 74 Configuring media parametrs 2 00125 cps enss bero edo RBOCIONR Ford dee xai iawn 74 Outgoing call ConNOUIAUON CCP m 76 Putting VolP lines intoa line POU asseoesamasuttieeRb 43d d REG RE Ren 76 Configuring telephones to access the VoIP lines 002 cece eee eee 78 Configuring a remote gateway sc cceee cee swede ed eee ea eee OR EX X RACER EE TB Coniiguring PSTN fallback Louisa aov dob ir ROPA bg e Spe doa o dpi n 80 Enabling ish falligek 2caieecerantadeens R004 42086 RACH RGSS er 4 SREE 81 Setting up tie VolP schedule ss5 ccna beans eee ARE ened oe denen e eas 82 COOMO OOE esa iaa Moog Meck tae dim a eR Heed Oba ded dd hun deodi dicii rud rdi dus 82 Creating destination codes for fallback lille 84 Activating tie VolP Seteduie 224 debe wA X Ende RR GAS SSE RE ed ados 87 Turning aud QoS ImODIIGI 022600 sqm resu Te woke seeds ERR EG Xx T4 RE 87 POIN Taba MEIGS uaa ao sm mea erinit irt Rede neo ie edes de eR OR ea 88 memng cal sno eon E FED EU 89 Assign o Target Nodo UP DM 4 4 bee oe ob qb RECAP e RC quer spes 89 Example coniguraton Sel lo sot uncos dAkeExe4 d UR OE I ate R
128. ephones to your calculations As soon as an IP telephone is registered it occupies an IP client whether it is active or not IP Telephony Configuration Guide 34 Chapter2 Prerequisites checklist Table 6 Keycodes Continued Prerequisites Yes No 5 c If you are using VoIP trunks do you need to activate MCDN features Note If MCDN is already configured on your system for private networking over land lines you do not need a separate MCDN keycode for VoIP trunks Business Communications Manager system configuration Several sections of the Business Communications Manager must be properly configured prior to activation of IP telephony Answer the questions in the following table to determine if your Business Communications Manager has been correctly configured Table 7 Business Communications Manager system configuration Prerequisites Yes No 6 a Is the LAN functioning correctly with the Business Communications Manager 6 b Is the WAN functioning correctly with the Business Communications Manager 6 c Have you determined the published IP address for the system Refer to Defining published IP address on page 35 6 d Has a dialing plan been created taking into account special considerations for IP telephony and private and public networking 6 e Do you want the system to auto assign DNs If no complete 6 f 6 f Have DN records been programmed for the
129. er buffer to improve quality when jitter is high Reduce packet errors Packet errors in intranets correlate to congestion in the network Packet errors are high because the packets are dropped if they arrive faster than the link can transmit Identify which links are the most used to upgrade This removes a source of packet errors on a distinct flow A reduction in hop count provides for less occurrences for routers and links to drop packets Other causes of packet errors not related to delay are as follows e reduced link quality e overloaded CPU saturation e LAN saturation e limited size of jitter buffer If the underlying circuit has transmission problems high line error rates outages or other problems the link quality is reduced Other services such as X 25 or frame relay can affect the link Check with your service provider for information Find out what the router threshold CPU utilization level is and check if the router conforms to the threshold If a router is overloaded the router is continuously processing intensive tasks Processing intensive tasks prevents the router from forwarding packets To correct this reconfigure or upgrade the router A router can be overloaded when there are too many high capacity and high traffic links configured on it Ensure that routers are configured to vendor guidelines Saturation refers to a situation where too many packets are on the intranet Packets can be dropped on improperly
130. ermine voice quality for IP telephones and trunks e Codecs on page 26 e Jitter Buffer on page 26 e QoS routing on page 27 IP Telephony Configuration Guide 26 Chapter Introduction Codecs The algorithm used to compress and decompress voice is embedded in a software entity called a codec COde DECode Two popular Codecs are G 711 and G 729 The G 711 Codec samples voice at 64 kilobits per second kbps while G 729 samples at a far lower rate of 8 kbps For actual bandwidth requirements refer to Determining the bandwidth requirements on page 121 where you will note that the actual kbps requirements are slightly higher than label suggests Voice quality is better when using a G 711 CODEC but more network bandwidth is used to exchange the voice frames between the telephones If you experience poor voice quality and suspect it is due to heavy network traffic you can get better voice quality by configuring the IP telephone to use a G 729 CODEC The Business Communications Manager supports these codecs e G 729 e G 723 e G729 with VAD e G 723 with VAD e GJll uLaw e GJll aLaw Jitter Buffer Voice frames are transmitted at a fixed rate because the time interval between frames is constant If the frames arrive at the other end at the same rate voice quality is perceived as good In many cases however some frames can arrive slightly faster or slower than the other frames This is called jitter and d
131. es a destination code of 3 e Under the Normal schedule the installer assigns Route 003 which uses line pool PRI A The absorb digits is set to All e Under the VoIP schedule the installer assigns Route 100 which uses the VoIP lines in line pool O The absorb digits is set to 0 The installer creates a destination code of 9 which will be used to access the local line pool for the local PSTN access lines e Under the Normal schedule the installer assigns the route created for the local PSTN access with absorb digits set to All e Under the VoIP schedule the installer assigns the route created for the local PSTN access with absorb digits set to All From the control set 2221 the installer dials FEATURE 873 and selects the VoIP schedule VoIP is now activated At this point the system is configured to make outgoing calls but it is not set up to receive incoming calls If there are no target lines set up the installer creates target lines for each DN or Hunt Group The Ottawa Business Communications Manager is now set to handle calls sent to and from a remote VoIP gateway However the Santa Clara Business Communications Manager must be set up before any calls can be made from that system IP Telephony Configuration Guide 94 Chapter5 Configuring VoIP trunks On Business Communications Manager Santa Clara This procedure details actions that the installer performs to set up the Business Communications Manager Santa Clara
132. etworking multiple systems 117 port settings 101 signaling method 104 system configuration prerequisites 34 using a gatekeeper 104 using firewalls 101 busy tone VoIP gateway progress tones 146 C call center networking multi locations 119 call chain network configuration 119 call progress tones 146 Call Signaling Local Gateway 106 call signaling modifying 104 calls gatekeeper examples 113 incoming configuration 89 making 95 media path redirection 144 capacity engineering link capacity 130 insufficient 130 Caution symbol 13 CDP network dialing plan 93 private network MCDN 116 changes to the intranet 134 checklist 29 clients media resources voice mail media resources WAN mediaresources 31 codecs IP Telephony Configuration Guide 172 Index defined 26 first preferred codec 144 for IP telephones 42 handling on network 122 types bandwidth 122 Unified Manager settings 53 comfort noise 140 computed load 156 computer IP telephony prerequisites 37 Conference Call 144 configure DN record 47 12050 Software Phone 64 IP server parameters 46 restart to 45 review information 48 Connecting to Server 47 contrast level IP telephones 49 control set setting the schedule 93 conventions and symbols 13 text 14 Coordinated Dialing Plan see CDP customize feature labels 59 D Danger symbol 13 Default gateway IP telephones 46 50 delay characteristics 156 endtoend 131 gathering statis
133. evel of QoS If it can the call proceeds as normal over the VoIP trunk If the minimum acceptable level of QoS is not met the call is routed over the second route through the PSTN line For PSTN fallback to work you must ensure that the digits the user dials will be the same regardless of whether the call is going over the VoIP trunk or the PSTN In many cases this involves configuring the system to add and or absorb digits This process is explained during the steps in Configuring routes on page 82 and Creating destination codes for fallback on page 84 For detailed information about inserting and absorbing digits see the Business Communications Manager 3 0 Programming Operations Guide Setting up PSTN fallback includes e Enabling PSTN fallback e Setting up the VoIP schedule e Configuring routes and dialing digits e Creating destination codes for fallback e Activating the VoIP schedule Turning on QoS monitor Enabling PSTN fallback To enable PSTN fallback 1 Open Services IP Telephony and click on H 323 trunks 2 Click the Fallback to Circuit Switched menu and select Enabled All or Enabled TDM only Enabled TDM only enables fallback for calls originating on digital telephones This is useful if your IP telephones are connected remotely on the public side of the Business Communications Manager network because PSTN fallback is unlikely to result in better quality of service in that scenario IP Telephony Configur
134. for the trunks overrides the telephone settings For IP telephones the Business Communications Manager supports both a law and mu law variants of the G 711 CODEC as well as the G 729 and G 723 CODECS e TheG 711 CODEC samples the voice stream at a rate of 64Kbps Kilo bits per second and is the CODEC to use for maximum voice quality Choose the G 711 CODEC with the companding law alaw or ulaw that matches your system requirements e The G 729 CODEC samples the voice stream at 8Kbps The voice quality is slightly lower using a G 729 but it reduces network traffic by approximately 80 e The G 723 CODEC should be used only with third party devices that do not support G 729 or G 711 e Codecs with VAD Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active P0993474 03 Chapter 3 Installing IP telephones 43 Choosing a Jitter Buffer A jitter buffer is used to prevent the jitter associated with arriving Rx voice packets at the IP telephones The jitter is caused by packets arriving out of order due to having used different network paths and varying arrival rates of consecutive voice packets The greater the size of the jitter buffer the better sounding the received voice appears to be However voice latency delay also increases Latency is very problematic for telephone calls as it increases the time between when one user speaks and when the user at th
135. g NetVision telephones on page 67 The applications that control these protocols on the Business Communications Manager provide an invisible interface between the IP telephones and the digital voice processing controls on the Business Communications Manager IP Telephony Configuration Guide 40 Chapter3 Installing IP telephones About Nortel Networks IP telephones The 12002 and 12004 telephones are hardwired to an internet connection They can be installed on any internet connection that has access to the network connected to the LAN or WAN of the Business Communications Manager The Nortel Networks 12050 Software Phone runs on any computer running Windows 98 Windows 2000 or Windows XP The computer must be connected to the LAN or WAN that the Business Communications Manager is connected to Configuring Nortel Networks i series telephones The configuration menus for the Nortel Networks i series IP telephones 12002 12004 12050 are under Services IP Telephony Nortel IP Terminals and Services Telephony Services System DNs Inactive DNs or Active set DNs once the telephone connects to the system This section contains the following information e Preparing your system for IP telephone registration on page 40 e Installing i series telephones on page 43 e Configuring the 12002 or 12004 telephone to the system on page 44 e Troubleshooting an IP telephone on page 48 e Configuring DHCP on page 49 e
136. gement systems have network planning modules that determine network flows These modules provide more detailed and accurate analysis because they can include correct node link and routing information They also help to determine network strength by conducting link and node failure analysis By simulating failures re loading network and re computed routes the modules indicate where the network can be out of capacity during failures Not enough link capacity If there is not enough link capacity consider one or more of the following options e Use the G 723 1 codec Compared to the default G 729 codec with 20 ms payload the G 723 1 codecs use 29 to 33 less bandwidth e Upgrade the bandwidth for the links P0993474 03 Efficient Networking 131 Other intranet resource considerations Bottlenecks caused by non WAN resources do not occur often For a more complete evaluation consider the impact of incremental IP telephony traffic on routers and LAN resources in the intranet where the IP telephony traffic moves across LAN segments that are saturated or routers whose central processing unit CPU utilization is high Implementing the network LAN engineering To minimize the number of router hops between the systems connect the gateways to the intranet Ensure that there is enough bandwidth on the WAN links shorter routes Place the gateway and the LAN router near the WAN backbone This prevents division of the constant bit rate IP
137. get lines for a range gt of telephones Refer to the Business Communications Manager 3 0 Programming Operations Guide for detailed information about using the wizard In Unified Manager open Services Telephony Services System DNs Under the Active Set DNs or under the Inactive DNs if you are preconfiguring DN records choose the DN record of the telephone where you want the line to be directed Choose Line Access Line assignment and click the Add button IP Telephony Configuration Guide 90 Chapter5 Configuring VoIP trunks 4 Enter the number of an available target line 241 492 Add Line assignment Click the Save button Click on the line number you just created and ensure that you have the line set to Ring Only if the telephone has no line buttons set for the line or Appearance and Ring if you are adding this to a DN that has line keys or which will be using SWCA keys Go to Services Telephony Services Lines Target Line Target line number from step 4 gt Click on the Trunk line data heading In the CLID set field enter the DN Target lines Distinct rings in use Line 241 4 Hue Line 242 a None Q Line 243 Distinct ring None v o General 4i CLID set DN 2243 vj Pwo wince cl icnla me gt 1a e at This allows the caller ID to display at the set before the call is answered 10 Click the key beside Trunk line data 11 Click on Received number 12 In the
138. h requirements on full duplex links llle 125 LAM engineering examples uus sede recommend ger eee oe Ree reas ame 126 P0993474 03 Table of Contents 7 WAN DOREM aucqiakoc dba 306 Rios Hee sd Eo eye bre om PROP Ra m hh 126 QoS Monitoring Bandwidth Requirement saana eee eee 127 Additional feature configuration 0 0 0 nee 127 Setting Non linear processing 2 iw sass sere ees bh e dol D aes XI RES C 128 Determining network loading caused by IP telephony traffic 0 128 nec pc 130 MST SOON TNR Capacity SED 130 Other intranet resource considerations 0 02 c eee eee 131 Implementing the network LAN engineering 00 eee eee eee eee 131 Further nebyoK analisi odere S pRCXTYATT TEENS Ebea SR PEAYPSas UwR SRG OSS Gwe 131 Componente ol delay 4 ccc ac icwarccarrdewe scent eraa REEE hee Rape o eee n 131 Hodube DRE Ol 1s naccsdopok a nakigun Liebe c e Bo cA Gp do RR C weed 132 FUE BOLD EQUI adds saexepieobpEaqsU eed psc pid eodd apu qa bag eid 132 Adjust the jitter buffer SEE d 33 aX eyed Ru eds CR C XR RRS 133 Reduce packet errors 2 ud iin wie Xe didt deae Xd e PR DR nba Ode o del doa ade 133 GU DHDDEC D T HR 134 Post installation network measurements 0 00 c eee ee eet 134 Appendix B Sience COMPICSTION ois oe esas x a RU E Rcs RE x Rcx uos E eee ews 135 Silence Compression on Half Duplex Links 0 00 cee eee 136 S
139. he central Business Communications Manager Business Communications Manager to IP telephones The system shown in the following figure allows home based users or Call Center agents to use the full capabilities of the Business Communications Manager including access to system users applications and PSTN connections This system does not require VoIP trunk configuration This system functions in a similar manner to the system described in Multi location chain with call center on page 119 This system is less expensive and on a smaller scale However it does not offer PSTN fallback Figure 41 Connecting to IP telephones System telephone Central Office i2050 Software Phone Intranet VoIP trunk i2004 Home based users telephone or Call Center agents Setting up a remote based IP telephone To set up this system 1 a W N Ensure that each remote user has a network connection capable of supporting VoIP traffic such as DSL or cable On the Business Communications Manager set up the system to support IP telephones At the remote location install and configure an IP telephone Register each telephone and provide it with a DN Set up the DN record with the required lines and services P0993474 03 121 Appendix A Efficient Networking This section provides information about making your network run more efficiently e Determining the bandwidth requirements
140. ications Manager is designed to interoperate with Radvision ECS 2 1 0 1 and CSE 1000 gatekeepers As part of this the Business Communications Manager supports both Direct GatekeeperResolved and Routed GatekeeperRouted call signaling in this mode of operation Note that if the call signaling method is changed the Business Communications Manager must be restarted before it functions properly Refer to Using a gatekeeper on page 103 for specific configuration instructions Asymmetrical media channel negotiation By default the Business Communications Manager IP Telephony gateway supports the G 729 codec family G 723 1 G 711 mu law and G 711 A law audio media encoding Because NetMeeting does not support the H 323 fastStart call setup method NetMeeting can choose a different media type for its receive and transmit channels However Business Communications Manager IP Telephony gateway does not support calls with different media types for the receive and transmit channels and immediately hangs up a call taken with asymmetric audio channels In this case the party on the Business Communications Manager switch hears a treatment from the switch normally a reorder tone The party on the NetMeeting client loses connection To solve this problem in NetMeeting under the Tools Options Audio Advanced check Manually configure compression settings and ensure that the media types are in the same order as shown in the Business Communications Manager
141. ields 107 Predefined Endpoints Properties settings 109 receive fallback threshold 154 receive path 124 receive threshold 78 88 92 recording routes 156 register IP telephone 40 IP telephones 47 Registration Disabled 48 Registration IP Radvision 109 Registration TTL Local Gateway 107 relocating IP telephones 63 Keep DN alive 63 remote access VoIP trunks 97 remote gateway activating QoS monitor 87 configuring 78 destination digits 80 MCDN networking 115 network example 92 VoIP trunks 78 remote routers setting up 146 remote system VoIP trunks 73 resource assessment prerequisites 31 router Business Communications Manager QoS support 161 intranet resource considerations 131 IP telephones 46 links to virtual circuits 122 network prerequisites 29 number of hops 131 port settings 103 Traceroute 141 routes full duplex link 129 recording 156 site pairs 156 routing and hop count 132 asymmetrical 134 P0993474 03 Index 179 delay issues 134 instability 134 network example 93 PSTN fallback 82 VoIP trunks 82 S S1 Action 46 SIIP 46 S1 Port 46 S1 RETRY Count 46 S2 Action 46 S2IP 46 S2 Port 46 S2 RETRY Count 46 schedule activating VoIP schedule 87 control set 93 destination codes 85 PSTN fallback 82 service setting manual 82 VoIP network example 93 SCNFallback 105 106 Scope status 50 serial cable NetVision 69 serialization delay 131 SERVER NO PORTS LEFT 48 server parameters 46 SE
142. ilence compression on Full Duplex Links leslssleeel In 138 eve C UEUUTBRMEBRETT 140 Appendix C Network performance utilities llle 141 PUB a aA ahh ns Rosie abaco Rott Eug de qs Bore gini ode qc UE E d p D uk e uc aude 141 NIS UE 5325259 HIE S E EE cay E qe OTe eS M IIR Tq qup dips 141 IN iret 1L ROUTE pay ee PRU ROLE DR RON ete infe Dos a odios dpa c Ri ood 142 Appendix D Lot ioo ro fem 143 Speech path setup metliD s uiuo sees pee ohh re ek Rec em kom dea RR Ree ame 144 Media path rediocton D deb iers ardt etn rai tiera beetiegdas 144 R O Socgqeqese qud NP Geena page ee PIE TEREDIPEPS TEES PIENO 145 Asymmetrical media channel negotiation 0 eee 145 N feedback busy SISION sed do aa eoe dition ence hike due eed UR de E aoe hese 146 Setting up Remote Routers for IP Telephony Prioritization llle 146 Creating an outbound traffic filter iusces reed ER 3 RB xv RR ER RES 146 Sample criteria ranges and actions for UDP filtering llus 147 Using VLAN on the nebWOrK iucecasouescusxker Rack E x ACE Coded CoRR de ERE Vom io 148 Choosing DAG Or VLAN io 2325 9eGa4 4 RR EX GU GRR eS ee EGG ER RUE 148 Specifying the site specific options for VLAN 20 eee eee eee ees 149 IP Telephony Configuration Guide 8 Table of Contents Symbol NetVision telephones isseaassrenk dea hh ER eU eee ES eRe EE ERS SOR RS 150 Software interoperability restrictions and limitations
143. ils Adding a NetVision record in the Unified Manager Follow these steps to preconfigure an H 323 Terminals record for each handset you install 1 Inthe Unified Manager open Services IP Telephony and click on H 323 Terminals The H 323 terminal list appears 2 On the top menu click Configuration and then click Add Entry The H 323 Terminal List dialog appears Figure 16 H 323 Terminal list dialog H 323 Terminal List Name Format Unique across first 7 chars DN p Password 1534 ede o TT Status Se EST Save Cancel 3 Use the information in the table below to set up your NetVision handset IP system record Table 15 H 323 Terminal list Field Value Description Name lt alphanumeric gt This is the name for the handset This name must have unique characters for at least the first seven digits Note This is the same name that you will enter in the Nortel NVPA configuration record for the User Name of the handset This name must be unique within the first seven characters for each handset and can be a maximum of 10 characters P0993474 03 Chapter4 Installing NetVision telephones 71 Table 15 H 323 Terminal list Continued Field Value Description DN DN number or This is the assigned DN for this handset If you want the system to 0 dynamically define a DN enter O zero Note This field cannot be left blank Password numeric
144. ions Manager is to be used as a firewall NAT device do the firewall rules fit within the 32 input rules and 32 output rules that the Business Communications Manager provides 2 g A hub based core will not have suitable performance for IP Telephony Does the network use a non hub solution at its core Network assessment The following table questions are meant to ensure that the network is capable of handling IP telephony and that existing network services are not adversely affected Table 3 Network assessment Prerequisites Yes No 3 a Has a network assessment been completed 3 b Has the number of switch hub ports available and used in the LAN infrastructure been calculated 3 c Does the switch use VLANs If so get the VLAN port number and ID 3 d Have the used and available IP addresses for each LAN segment been calculated 3 e Has DHCP usage and location been recorded 3 f Has the speed and configuration of the LAN been calculated P0993474 03 Chapter 2 Prerequisites checklist 31 Prerequisites Yes No 3 g Has the estimated latency values between network locations been calculated 3 h Have the Bandwidth CIR utilization values for all WAN links been calculated 3 1 Has the quality of service availability on the network been calculated Resource assessment Answer the questions in the following table to determine if you h
145. ird Route None x Sched 6 wild cards Absorbed length All x n Change Use Route to the route you configured for your VoIP line Set the Absorbed length to 0 Note In this case the destination code and the gateway destination digit are the same Note that you can add up to three alternate routes Note If the destination code is different from the remote gateway destination digits and you entered an External into the route record set the absorbed length to the number of digits in the destination code The system will dial out the External you entered in front of the rest of the number that the user dialed Or you can use the destination digits as part of the destination code and set the absorbed length to 1 to absorb the destination code but still dial the destination digits so the system can find the gateway IP Telephony Configuration Guide 86 Chapter5 Configuring VoIP trunks 7 On the navigation tree under the destination code schedule click Normal The Normal schedule appears It contains the same two fields as shown in the figure above a Change Use Route to the route you configured for your PSTN fallback line the line to the other system b Set the Absorbed length to All Figure 25 Normal schedule routing information e Del All Destination codes lt 5A Normal 5A Normal Q 5A Use route Route 001 M 9 Schedules Absorbed length All v Night
146. ister Open the Configuration menu or right click anywhere on the listing for the terminal to bring up the menu as shown in the next figure Deregister DN from Configuration menu View Help Summary General IP Terminal Status Telephony Features List IP Terminal Status Offline Modify parameters Default 2432 Offline Default 2433 Offline i20 Default Force firmware download Reset Hot Desking Password 5 Click Deregister DN 6 Reregister the telephone as described in Configuring the i2002 or i2004 telephone to the system on page 44 j Warning Once this feature is activated all active calls are dropped P0993474 03 Chapter 3 Installing IP telephones 63 Moving IP telephones IP telephones retain their DN when they are moved to a new location on the same subnet The following instructions apply to Nortel IP telephones To move an IP telephone without changing the DN 1 2 3 4 Disconnect the power from the IP telephone or 3 port switch Disconnect the network connection At the new location reconnect the network cable and the power connection If the new location is on a different subnet you will need to make the appropriate changes to the telephone IP addressing However do not change the S1 IP address or the S2 IP address Note If your network is using partial DHCP reconfiguration is not require
147. istics of the codecs Each codec delivers a different MOS for the same network quality IP Telephony Configuration Guide 162 Quality of Service Network monitoring The VoIP Gateway network monitoring function measures the quality of service between the local and all remote gateways on a continuous basis The network monitoring function exchanges UDP probe packets between all monitored gateways to collect the network statistics for each remote location All the packets make a round trip from the Sender to Receiver and back to the Sender From this information you can calculate the latency and loss in the network for a distinct location Note 1 Quality of Service monitoring is supported only on Business Communications Manager MI with ITG card and 120xx Note 2 The Quality of Service threshold is configurable per remote gateway Note 3 Fallback starts for all new originating calls if the QoS of any monitored gateway is below its threshold Note 4 The fallback decision is made only at the originating gateway using the QoS thresholds monitored at the originating gateway for the destination gateway VoIP Gateway allows for manual configuration of QoS thresholds depending on the customer preference between cost and voice quality Quality of Service parameters Quality of Service depends on end to end network performance and available bandwidth A number of parameters determine the VoIP Gateway QoS over the data network The V
148. kbone The major transmission path of a network handling high volume high density traffic bandwidth A measure of information carrying capacity available for a transmission medium shown in bits per second The greater the bandwidth the more information sent in a given amount of time bridge LAN equipment providing interconnection between two networks using the same addressing structure A bridge filters out packets that remain on one LAN and forwards packets for other LANs codec Equipment or circuits that digitally code and decode voice signals Software that provides compression decompression algorithms for voice traffic over IP networks and VoIP trunks communications protocol A set of agreed upon communications formats and procedures between devices on a data communication network data communications Processes and equipment used to transport signals from a data processing device at one location to a data processing device at another location default gateway For IP telephony this refers to the router that closest to the IP telephone DS30 split This term refers to the allocation of media resources by the media services card MSC on the Business Communications Manager The default setting is a 2 6 split meaning that DS 01 and DS 08 are automatically used internal media processing including IP telephony If you plan to have a maximum number of IP telephones you may need to set your system so that it uses DS30
149. ks 74 LAN Business Communications Manager function 34 engineering examples 126 implementing the network 131 Published IP address 35 late packets 157 latency jitter buffer 43 line pool access codes 77 adding to DN record 78 network example 92 VoIP trunk routing 82 VoIP trunks 76 IP Telephony Configuration Guide 176 Index lines VoIP trunks default 76 link capacity insufficient 130 capacity system engineering 130 delay 132 full duplex bandwidth requirements 125 half duplex bandwidth requirements 124 local gateway Alias Names 106 Call Signaling 106 Fallback to Circuit Switched 105 Gatekeeper IP 106 Gateway Protocol 107 Registration TTL 107 Locating Server 47 MI also see Meridian 1 and M1 ITG MI TG 74 MI ITG also see M1 defined 23 gateway type 115 Interoperability 143 payload size 144 profile agreement 144 making calls VoIP trunks 95 Maximum cell rate MCR 122 MCDN gateway type 115 MI ITG requirements 116 over VoIP 80 115 PRI fallback 116 remote gateway 115 measurements post installation 134 Measuring Intranet QoS 155 media channels asymmetrical negotiation 145 media parameters VoIP trunks 74 Media path redirection 143 media resources prerequisite 31 menu list feature 900 56 Meridian 1 also see M1 MI TG 74 MCDN networking 115 profile 144 monitoring the network 134 MOS range 154 moving IP telephones 63 Keep DN alive 63 mu law 145 multi locations ne
150. l In this method call signaling is directed through the gatekeeper e Call Signaling i Circuit Switched SCNFalback Call Signaling CallSignaling Direct Gatekeeper IP GatekeeperRouted GatekeeperResolved Alias Names CallSignalina Gatekeeper IP ip address If GateKeeperRouted or GateKeeperResolved are selected under Call Signaling type the IP address of the machine that is running the gatekeeper Alias Names lt alphanumeric gt Refer to example below If GateKeeperRouted or GateKeeperResolved are selected under Call Signaling type one or more alias names for the gateway One or more alias names may be configured for a Business Communications Manager Alias names are comma delimited and may be one of the following types e E 164 numeric identifier containing a digit in the range 0 9 Identified by the keyword TEL e NPI TON also referred to as a PartyNumber alias Similar to E164 except that the keyword indicates the NPI numbering plan identification as well as the TON type of number Identified by one of the following keywords PUB Public Unknown Number PRI Private Unknown Number UDP Private Level 1 Regional Number UDP CDP Private Local Number CDP Refer to Notes about NPI TON aliases on page 107 e H323ldentifier alphanumeric strings representing names e mail addresses etc Identified by the keyword NAME P0993474 03
151. lay until a port becomes available and the telephone is powered down then powered up To obtain more ports you may need to install additional VoIP keycodes See the Keycode Installation Guide Invalid Server Address The S1 is incorrectly configured with the IP address of a Business Communications Manager network adapter other than the published IP address IP Address conflict The telephone detected that a device on the network is currently using the IP address allocated to the telephone Registration Disabled The Registration on the Business Communications Manager is set to OFF SERVER UNREACHABLE Check that you have entered the correct Netmask and gateway IP addresses RESTARTING If the settings are correct contact your system administrator NEW SET The telephone has not been connected to the Business Communications Manager before and must be registered Note To see the configuration information for a telephone connected to the Business Communications Manager When the telephone is not on a call press the key bottom right corner of the telephone followed by the Ce key next to the key The display will automatically scroll through the configuration settings eg key play y g g g To see the Codec data for a telephone while it is on a call Press the key followed by the Ce key Other troubleshooting tips Here are a few possible issues you may encounter plus a description of what ma
152. lback you must assign the line pool you create in this procedure to a route and then you need to specify a destination code Refer to Configuring PSTN fallback on page 80 P0993474 03 Chapter 5 Configuring VoIP trunks 77 To put your lines into a line pool 1 In Unified Manager click on the keys beside Services Telephony Services Lines VoIP lines Enabled VoIP lines 2 Click on Line XXX where XXX is the line number for the VoIP trunk you want to put in the line pool 3 Click on Trunk Line Data The Trunk Line Data screen appears as shown in the figure below Figure 18 Trunk Line data CT Line 001 Trunk line data te Del All r Line 001 Trunkdine data All VoIP lines j 9 Line 001 Trunk type ap i UE General Trunkiline data 9 Restrictions Line type Pool H x Line restric Prime set DN 2221 I Ramnte recs In the Line type field set a line pool that is not used by any non VolP lines Repeat this procedure for as many trunk lines as you have keycodes for You can use the same line pool for all VoIP lines 6 On the navigation tree click the keys beside General Settings Access Codes and Line Pool Codes Note Set up an access code for the line pool only if you are NOT planning to use PSTN fallback If you intend to use PSTN fallback you must assign the line pool you create in this procedure to a route and then you need to specify a destination code Refer to
153. ld it will trigger fallback to PSTN For information about setting up the system to use QoS and fallback to PSTN see Configuring PSTN fallback on page 80 Bandwidth required for QoS monitor There are a total of 25 monitoring packets traveling in each direction every 15 seconds Each of monitoring packages has 88 bytes in IP layer These monitoring packets are equally spaced out in the 15 second interval For example if there are two Business Communications Managers BCM A and BCM B connected to each other with QoS Monitoring enabled In every 15 seconds there are 25 monitoring packages going from BCM A to BCM B and then back to BCM A Similarly the same occurs from BCM B to BCM A back to BCM B In other words in this case the overhead in IP layer caused by these monitoring packets is about 2x25x88 152 293 bytes second 2346 bits second in one direction Quality of Service Status The QoS Status displays the current network quality described as a Mean Opinion Score MOS for each IP destination A pull down menu allows the administrator to view the MOS mapping The table below shows a sample QoS Monitor Table 18 QoS status G 723 1 6 3 G 723 1 5 3 G 729 G 711 kbit s kbit s QoS IP Monitor Tx Rx Tx Rx Tx Rx Tx Rx 47 192 5 2 Enabled 4 50 4 50 4 00 4 30 4 75 4 70 4 80 4 90 47 192 5 6 Disabled N A N A N A N A N A N A N A N A Note For the QoS monitor and PSTN fallback to function
154. lder model 12004 telephones This switch allows the telephone and computer to connect to the same network connection For more information consult the 12004 and the 3 way switch documentation The 12002 and newer models of the 12004 telephone have an adapter that replaces the requirement for this switch IP Telephony Configuration Guide 44 ChapterS3 Installing IP telephones Connecting the i2002 or i2004 telephone Follow these steps to connect an 12002 or 12004 telephone 1 2 Connect one end of the handset cord to the handset jack on the telephone base Connect the other end of the handset cord to the handset Connect one end of a Cat 5 line cord with RJ45 connectors to the line cord jack on the telephone base Connect the other end of the line cord to the Ethernet connection or to the 3 way switch connector Note Newer i20XX terminals have a 3 way switch built into the telephone Refer to the gt installation card that comes with the telephone for specific directions 5 6 Plug the AC Power adapter into the base of the telephone and then plug the adapter into the AC outlet Go to Configuring the 12002 or i2004 telephone to the system Configuring the i2002 or i2004 telephone to the system Configuring IP telephones involves two processes If DHCP Distributed Host Control Protocol service on the Business Communications Manager is active or the Customer DHCP server has been configured to hand out
155. le by silence compression is available for lower priority data applications that can tolerate increased delay and jitter Figure 51 Two calls on a full duplex link with silence compression Conversation 1 Buenos noches Juan Muy bien y tu x D Rx Hofa Isabella em O esias Tx Hello Fred This is Susan Do you have a minute Rx Fred here fp Hi Sure Conversation Bandwidth used EN Channel Link max Y Channel Link max Rx channel Bandwidth HE EM Comfort noise To provide a more natural sound during periods of silence comfort noise is added at the destination gateway when silence compression is active The source gateway sends information packets to the destination gateway informing it that silence compression is active and describing what background comfort noise to insert The source gateway only sends the information packets when it detects a significant change in background noise P0993474 03 141 Appendix C Network performance utilities There are two common network utilities Ping and Traceroute These utilities provide a method to measure quality of service parameters Other utilities used also find more information about VoIP Gateway network performance Note Because data network conditions can vary at different times collect performance data over at least a 24 hour time period This section also describes the Sniffer utility
156. lect the Codecs in the order in which you want the system to 2nd Preferred Codec G 711 uLaw attempt to use them 3rd Preferred Codec G 711 aLaw 4th Preferred Codec G 729 1st Preferred Codec Ges 2nd Preferred Codec G 729 VAD G 723 VAD 3rd Preferred Codec 4th Preferred Codec G 711 aLaw Performance note Codecs on all networked Business Communications Managers must be consistent to ensure that interacting features such as Transfer and Conference work correctly Refer to Codecs on page 26 Silence Compression Enabled The silence compression identifies periods of silence ina Disabled conversation and stops sending IP speech packets during those periods In a typical telephone conversation most of the conversation is half duplex meaning that one person is speaking while the other is listening If silence compression is enabled no voice packets are sent from the listener end This greatly reduces bandwidth requirements G 723 1 and G 729 support silence compression G 711 does not support silence compression Silence Compression Jitter Buffer Voice Performance note Silence Compression on all networked Business Communications Managers and ITG systems VAD setting on ITG systems must be consistent to ensure that interacting features such as Transfer and Conference work correctly As well the Payload size on the ITG must be set to 30ms IP Telephony Configuration Guide 76 Chapter5 Configuring VoIP tru
157. les Feature code New label Feature code New label 60 Gen Page 521 SW Call 1 610 Pg Every 522 SW Call 2 61 Zone digit from 1 9 523 SW Call 3 62 Speak Pg 524 SW Call 4 630 Speak All 525 SW Call 5 Note Line names are defined when you configure the line and can be changed through the Lines menus Download firmware to a Nortel IP telephone Firmware is the software stored in the telephone When the Business Communications Manager is upgraded with a new IP telephone firmware load this firmware load will automatically be downloaded into the IP telephones when they next connect to the Business Communications Manager You can use the Force firmware download option under the Configuration menu Nortel IP Terminals to force immediate download to a telephone You would do this in situations where you suspect that a particular telephone has corrupted firmware P0993474 03 Chapter 3 Installing IP telephones 61 Follow these steps to force a firmware download to a telephone 1 Inthe Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary appears 2 Click on the IP Terminal Status tab Select the IP telephone that you want to download firmware to Open the Configuration menu or right click anywhere on the listing for the terminal to bring up the menu A Ce Performance Fault Modify parameters Deregister DN
158. lients Net Meeting is an application available from Microsoft which uses the H 323 protocol To use Net Meeting 1 Install Net Meeting on the client computer 2 Inthe Tools menu click Options The options dialog appears Figure 31 NetMeeting options uls microsoft com P0993474 03 Chapter 5 Configuring VoIP trunks 99 3 Click Advanced Calling The advanced Calling Options dialog appears Figure 32 NetMeeting advanced options Advanced Calling Options Log on Using my account name ECOUL HOMNE A Log on using my phone number EE 4 Under Gateway settings select the Use a gateway option In the Gateway field type the published IP address of the Business Communications Manager Click OK Add a remote gateway to your system as explained in Configuring a remote gateway on page 78 When prompted for the IP address of the remote gateway type the IP address of the client computer Repeat this procedure for every NetMeeting client you want to set up IP Telephony Configuration Guide 100 Chapter5 Configuring VoIP trunks Quality of Service Monitor The Quality of Service Monitor is an application that monitors the quality of the IP channels It does this by performing a check every 15 seconds The QoS Monitor determines the quality of the intranet based on threshold tables for each codec If the QoS Monitor is enabled and it determines that the quality of service falls below the set thresho
159. media parameters table The following table lists the names used by the Business Communications Manager local gateway table and the matching names in NetMeeting Table 35 Name comparison Business Communications Manager media parameters table MS NetMeeting G 723 1 6 3 Kbit s MS G 723 6400 bit s G 723 1 5 3 Kbit s MS G 723 5333 bit s G 711 u law CCITT u law G 711 A law CCITT A law IP Telephony Configuration Guide 146 Interoperability No feedback busy station The Business Communications Manager VoIP gateway provides call progress tones in band to the user If a busy station is contacted through the gateway the gateway plays a busy tone to the user However as NetMeeting does not support fastStart no speech path is opened to the user before the call connects Because of this the user on the NetMeeting station does not hear a busy signal from the gateway Setting up Remote Routers for IP Telephony Prioritization This section includes information about setting up earlier version of BayStack routers and how to set up a range of UDP as a high priority DiffServ Code Point DSCP Creating an outbound traffic filter To create an outbound traffic filter 1 o0 An 10 11 In the Configuration Manager window click Circuits and then click Edit Circuits The Circuit List window appears Select a circuit Click the Edit button The Circuit Definition screen appears with the
160. meters 47 565 138 52 stem Echo Cancellation Enabled w NLP x sources vices G 723 1 Data Rate 6 3 kbps X Telephony Services HIP Telephony a Reserved Media Gateway Codec G 711 v 3 Change the settings for the fields below as required for your system Table 5 IP terminals general record fields Field Value Description Echo Cancellation Enabled w NLP Enable or disable echo cancellation for your system Enabled Disabled Echo Cancellation Enabled w NLP v Enabled G 723 1 Data Rate Disabled Reserved Media Gateway Codec E Echo Cancellation selects what type of echo cancellation is used on calls that go through a Media Gateway NLP refers to Non Linear Processing P0993474 03 Chapter2 Prerequisites checklist 33 Table 5 IP terminals general record fields Continued Field Value Description G 723 1 Data Rate 5 3 kbps Choose the preferred data rate for the channel 6 3 kbps G 723 1 Data Rate 6 3 kbps Y 5 3 kbps ved Media Gateway Codec G 723 1 Data Rate selects what data rate is used for transmissions from the Business Communications Manager to an IP device when the G 723 1 family codec is used G 723 1 or G 723 1A This has no effect on any other codec The possible values are 5 3 kbps and 6 3 kbps Reserved Media G 711 Choose the preferred codec that you are using with your IP network Gateway Codec G 729 G 723 Reserved Media Ga
161. mmunications Manager PSTN fallback route System Intranet Pii rnm VoIP trunk i2004 telephone To set up this system 1 Make sure the M1 ITG meets the following requirements ITG 2 X 26 Rls25 30 or higher S W Packages 57 58 59 145 147 148 160 2 Ensure that the MI ESN programming CDP UDP is compatible For information on this refer to your M1 documentation 3 Onthe Business Communications Manager 3 0 Unified Manager Set up outgoing call configuration for the VoIP gateway Set up a remote gateway for the Meridian 1 Ensure the dialing rules CDP or UDP are compatible with the M1 For information on CDP and UDP refer to the Programming Operations Guide Configure the PSTN fallback and enable QoS on both systems If target lines have not already been set up configure the telephones to receive incoming calls through target lines MCDN functionality on fallback PRI lines To be able to use MCDN functionality over PRI fallback lines set up e Check MCDN PRI settings on the M1 For information on this refer to the M1 documentation e Ensure SL 1 MCDN keycodes are entered on the Business Communications Manager 3 0 and the PRI line is set up for SL 1 protocol P0993474 03 Chapter6 Typical network applications using MCDN 117 For a detailed description of setting up fallback refer to Chapter 5 Configuring VoIP trunks on page 73 Networking multiple Business Commu
162. n refer to the documentation for the application A OO N LR Open the Radvision application In the left frame click the Calls button Ensure the following fields are set Table 21 Radvision Calls screen required settings On the viaIP Administrator screen select the Settings tab then click on the Basics button Beside the Who can register field choose Everyone Field Value Description Accept calls check box Box must be checked Routing Mode Direct Set to Direct Setup Q 931 not supported Call Control H 245 Nortel recommends that you always use Direct mode Check that call is active every check box Leave box UNCHECKED Enabling this feature will result in dropped calls 5 Inthe left frame click the Advanced button Ensure the following fields are set Table 22 Radvision Advanced screen required settings Field Value Description Check that the endpoint is check box Leave box checked online every This setting controls the intervals when Radvision checks if the Business Communications Manager is still on line Enable TTL check box Box must be checked This is the only mechanism currently supported that allows the gatekeeper to determine if the end point the Business Communications Manager is active Force Direct for Service Calls check box Check this box if you selected the Routing Mode Direct on the Calls screen P0993474 03
163. n environments where the initial choice may already be in use by a different vendor Pick only one TYPE byte Length 1 octet variable depends on the message content Data length octets e ASCII based e format VLAN A XXX YYY ZZZ where VLAN A uniquely identifies this as the Nortel DHCP VLAN discovery A signifies this version of this spec Future enhancements could use B for example ASCII comma is used to separate fields ASCII period is used to signal end of structure XXX YYY and ZZZ are ASCII encoded decimal numbers with a range of 0 4095 The number is used to identify the VLAN Ids A maximum of 10 VLAN Ids can be configured NONE means no VLAN default VLAN The DHCP Offer message carrying VLAN information has no VLAN tag when it is sent out from the DHCP server However a VLAN tag will be added to the packet at the switch port The packet will be untagged at the port of the IP phone IP Telephony Configuration Guide 150 Interoperability Symbol NetVision telephones In order to make calls between Symbol telephones and Business Communications Manager 3 0 each must be configured to have at least one common codec The following codecs are supported by the NetVision telephones e G 711 u law e G711 A law e G 729 Annex A and Annex B Software interoperability restrictions and limitations The following tables provide a brief overview of the IP telephony H 323 compatibility issues incl
164. n rate for G 729 is 24 8 kbit s per call or 99 2 kbit s in each direction for all four calls In other words in order to support four G 729 calls the WAN link must have at least 99 2 kbit s of usable bandwidth in each direction P0993474 03 Efficient Networking 127 The average bandwidth for each call is 12 4 kbit sec per channel or 49 4 kbit s for all four calls for each channel Low priority data applications can make use of bandwidth made available by silence suppression Figure 43 Peak traffic WAN link PPP B W Silence NoSP Suppression peak peak Avg kbit s kbit s kbit s G 729 10 8 kb s 20 24 8 12 4 30 QoS Monitoring Bandwidth Requirement The VoIP Quality of Service QoS Monitor periodically monitors the delay and packet loss of IP networks between two peer gateways e g Business Communications Manager to Business Communications Manager by using a proprietary protocol The main objective of the QoS Monitor is to allow new VOIP calls to fall back to the PSTN if the IP network is detected as bad in terms of delay and packet loss For more details about configuring QoS Monitoring refer to the Programming Operations Guide The monitoring packets are delivered at UDP port 5000 If you use QoS Monitoring in your gateway setting please refer to the following paragraph for a description of bandwidth requirement of QoS Monitoring There are a total of 25 monitori
165. n requirements 43 prerequisites 29 IP telephones 37 keycodes 33 MI ITG MCDN 116 network assessment 30 network devices 30 network diagram 29 resource assessment 31 system configuration 34 PRI MCDN fallback 116 private IP address 30 36 92 prompts IP telephones configuration 48 propagation delay 131 protocol IP Telephony Configuration Guide 178 Index link bandwidth requirements 124 125 remote gateway 78 PSTN fallback 76 80 activating VoIP schedule 87 configuring 80 destination codes 84 dialed digits 82 enable 81 MCDN networking 116 mean opinion score 163 PRI line 91 scheduling 82 public IP address 30 36 92 Published IP address choosing 36 determine which IP address to use 36 IP telephones 46 network example 92 setting 35 VoIP trunks 35 QoS analysing 159 Business Communications Manager gateway router support 161 defined 27 implementing in IP networks 159 MCDN networking 116 measuring intranet 155 MOS range qualitative scale 154 objectives 153 parameters 122 setting 153 status 100 QoS monitor activating 87 enabled 92 remote gateway 78 status display 100 updating data 100 qualitative scale QoS 154 Quality of Service Monitor see QoS monitor queuing delay 132 R RI determining link capacity 130 peak VoIP load 129 R2 determining link capacity 130 peak VoIP load 129 Radivision interoperability support 109 Radvision ECS 2 1 0 1 gatekeeper 108 mandatory f
166. nce The Gatekeeper is the final decisionmaker It will select the mode routed or resolved based on its configuration e GatekeeperRouted routes the Call Setup Channel and Control Channel through the ECS In ECS terminology this mode is called Call Setup Q 931 and Call Control h 245 e GatekeeperResolved routes the Call Setup Channel and Control Channel directly to the far end without ECS intervention In ECS terminology this mode is called Direct By using this method you will speed up you call setup time This is the recommended configuration for the Business Communications Manager ECS Configuration e Accept calls this must be enabled so that calls pass through the ECS Gatekeeper e Routing Mode it is recommended that you set this to Direct to minimize call setup time The Business Communications Manager also supports routing of Setup Q 931 and Call Control H 245 Important The Business Communications Manager does NOT support the second option the routing of Setup Q 931 The option Check that call is active every XXX seconds must be unchecked e Force Direct For Service Calls this setting on the Settings Advanced tab should be enabled if the ECS Gatekeeper has been configured to use Direct call routing ITG version 26 26 does not include support for gatekeeper interaction To be able to establish calls between Business Communications Manager 3 0 and ITG through a gatekeeper follow the configuration steps found in
167. ndicate network congestion If the packets time out then the remote device is unreachable The round trip time rtt is indicated by the time field So that the delay sample results match what the gateway experiences both the Ping host and target must be on a functioning LAN segment on the intranet Set the size of the Ping probe packets to 60 bytes to approximate the size of probe packets sent by IP telephony This determines if new calls need to fall back on the circuit switched voice facilities Notice from the Ping output the difference of rtt The repeated sampling of rtt allows you to receive a delay characteristic of the intranet To get a delay distribution include the Ping tool in a script which controls the frequency of the Ping probes which timestamps and stores the samples in a raw data file The file can be analyzed by the administrator using spreadsheets and other statistics packages The installer can check if the intranet network management software has any delay measurement modules which can cause a delay distribution measurement for specific site pairs IP Telephony Configuration Guide 156 Quality of Service Delay characteristics vary depending on the site pair and the time of day The evaluation of the intranet includes taking delay measurements for each site pair If there are important changes of traffic in the intranet include some Ping samples during the peak hour For a more complete evaluation of the int
168. ne ein See Eee LEAS eE A EROS AR EE 59 Download firmware to a Nortel IP telephone 2 000 eee ees 60 Deregistering DNs for IP telephones 2 0 oc ke RR RR 62 Moving IF lip Me T CCP MTMTUTMTT 63 oup DP SC caduta vaccis e ao e e Rp a ede DE cda did e abaco ood ata 63 Configuring the Nortel Networks i2050 Software Phone 220000 ee eee 64 P0993474 03 Table of Contents 5 Chapter 4 Installing NetVision telephones sereni 67 N ervision GOnhDgeliVIDy uuessexesexeusexeesusekaeaewesdaaxteadaxesd aqu gd beeen 67 DUGDBS DUM ad pagar RAE VIRG Ced b SE dS SOR CEU A dbi VR bd xd Nd 68 qs oo MP PETITUM 68 Handset and call TUGoHOtiks uui si ded ERST E Eh d Rx e d Een d node ae eee 68 Coniguring Nefvision records cscs Leser bee eo ba purae S Redondo pac peur bs De Rd 68 Gathering system information before you start lille 69 Assigning H 323 Terminals records i 22445 esas sees debe RR REOR RR eens 69 pr Mer LC Mq cc cr TT 69 Adding a NetVision record in the Unified Manager 00 eee 70 Testing the handset functions V espocesxaco e yw Scand ee geb RO doen e RC otn Y1 Updating the H 323 terminals record sser eue Vw TERI TREQEPEQETAMO T TF eso ES 71 Changing a handset Maie cccscc deers dee eedev econo de xa ye re REY XE EU RR Rd r Changing the DN record of a handset ssseusu uus ew rinni xw RO Dx RR m oan bbw Ta Deleting a NetVision telephone from the syst
169. nes e Configure the PSTN fallback and enable QoS on both systems 4 Reboot each system This system uses fallback to PSTN so calls can be routed across the PSTN connection if VoIP traffic between the Business Communications Manager systems becomes too heavy A similar system is shown below except that only one of the Business Communication Managers has a line to the PSTN network In this case all public calls from both systems are funneled through the system with the PSTN connection and all communication between the systems occurs over IP trunks To facilitate this system you need to ensure that the routing codes on the non PSTN system point to the system connected to the PSTN and then to the PSTN On the PSTN connected system the system and routing codes must be configured to recognize and pass public calls from the other system out into the PSTN network This also means that if the VoIP trunks are inaccessible between the systems there is no provision for a fallback route Figure 39 Routing all system wide public calls through one Business Communications Manager Head Office Warehouse Business Business System Communications Communications System telephone Manager Manager telephone soos 33 IED Intranet VoIP trunk i2050 Software Phone i2004 telephone i2004 telephone remote i2004 The Programming Operations Guide provides a
170. ness Communications Manager but it is a useful tool for diagnosing network functionality It provides origin destination and header information of all packets on the data network P0993474 03 143 Appendix D Interoperability This section discusses interoperability between the Business Communications Manager and other networks including e Speech path setup methods on page 144 e Media path redirection on page 144 e Gatekeeper on page 145 e Asymmetrical media channel negotiation on page 145 e Setting up Remote Routers for IP Telephony Prioritization on page 146 e Using VLAN on the network on page 148 e Symbol NetVision telephones on page 150 e Software interoperability restrictions and limitations on page 150 Business Communications Manager 3 0 IP Telephony adheres to the ITU T H 323v2 standards Such endpoints include the Nortel Networks M1 ITG and Microsoft NetMeeting As well the Business Communications Manager is backward compatible and interoperates with the Nortel Networks 12002 12004 telephones and 12050 Software Phone and with the Symbol NetVision IP Phones The following table summarizes this information Table 32 Business Communications Manager 3 0 Product Interoperability Summary Vendor Product Version Nortel Networks Business Communications Manager 2 0 or greater Nortel Networks i2002 12004 3002B20 or greater
171. ng Nortel Networks i series telephones lees 40 Preparing your system for IP telephone registration llle 40 Siero DoE Eres aAa T TEE oi eica RE care meena E e REC E dfe 42 Choosing aJitter Buler oi ceera cei da eee dob oe RE eo ri RARE AR RE Ru 43 Installing seres telephones 2 ccc ies has arn ph AUR ee eh RR RR XD TREE 43 Deore Weill serris srsti reti satan gee Y REEF Cadba se Raw a news 43 VSG a Spon SWIIGI iuauioa meds ehe Re E E ERR E G Uc sweet ange ae dd 43 Connecting the i2002 or i2004 telephone 0 00 44 Configuring the i2002 or i2004 telephone to the system llle An Registering the telephone to the system anaana eee eee eee 45 Configuring telephone settings sisi bdo ce En kee ee RR ROS eed 45 Troubleshooting an IP telephone saec ec mx mmm 48 Other Troubleshooting US usas 0600 e re oot OY V nee dd xr e HORE RR 48 COMMING BNO em CURES 49 IP telephony DACP fiBEGe sc iussus x eRk RE ped RR RR RE ERR ERR 50 Checking IP server Status cuc oie cheese itsur oeu Ed eau wed Rew ca a Un 52 Modifying IP telephone status settings liliis 53 Working with the features list ios reseao en RE eu hERERE RE TERRE RE ERR 55 Using the Services button to access features liliis 56 Using the Hot Desking feature iiuuse cusa m es Sea bv eee Y Ree wert ee ee eee e d 57 Notos about Mot Desking 2544480 4865 4454 4 RE PETAR E eA eee 58 Customizing feature labels ise esas a
172. ng packets traveling in each direction every 15 seconds Each of monitoring packages has 88 bytes in IP layer These monitoring packets are equally spaced out in the 15 second intervals For example if there are two Business Communications Managers BCM A and BCM B connected to each other with QoS Monitoring enabled then in every 15 seconds there are 25 monitoring packets going from BCM A to BCM B and then back to BCM A Similarly 25 packets go from BCM B to BCM A then back to BCM B In other words in this case the overhead in IP layer caused by these monitoring packets is about 2x25x88 152 293 bytes second in one direction Additional feature configuration This section contains additional information about configuring your network to run efficiently Setting Non linear processing on page 128 e Determining network loading caused by IP telephony traffic on page 128 e Implementing the network LAN engineering on page 131 IP Telephony Configuration Guide 128 Efficient Networking Setting Non linear processing Non linear processing should normally be enabled To set non linear processing 1 InUnified Manager open Services IP Telephony and click on H 323 settings The H 323 parameters appear in the right window 2 Click the Non linear processing drop down menu and select either Enabled or Disabled Determining network loading caused by IP telephony traffic At this point the installer or administrator h
173. nge the properties for Open the Configuration menu or right click anywhere on the terminal listing to open the Configuration menu From the menu select Modify parameters The IP Terminal Status dialog box appears as shown in the figure below IP Telephony Configuration Guide 54 Chapter3 Installing IP telephones Figure8 IP Terminal status dialog IP Terminal Status DN 2431 Read Only Field Status Offline Type i2050 IP Address IN 7A Codec Default v FAW Version IN 7A JitterBuffer Default v Terminal ID N HA 6 You can change the Codec or JitterBuffer settings for the terminal The table below describes the fields on this screen Table 13 IP Terminal Status fields Field Value Description DN Read only This is the DN record that is assigned to this terminal Status Read only This is the current status of the terminal Type Read only This is the type of IP telephone assigned to this record i2050 i2004 i2002 IP address Read only This is the IP address assigned to this telephone Codec Default Specifying a non default CODEC for a telephone allows you to G 711 aLaw override the general setting You might for example want to specify G 711 uL a low bandwidth CODEC g 729 for a telephone that is on a remote ub esis or busy sub net G 711 with VAD Refer to Choosing a codec on page 42 G 729 G 729 with VAD G 723 F W version Read only P0
174. nications Managers The system shown in the following diagram allows multiple offices with Business Communications Manager systems to connect across the company Intranet This installation allows for CallPilot to direct calls throughout the system Full toll bypass occurs through the tandem setup meaning that any user can call any DN without long distance charges being applied Users have full access to system users applications PSTN connections and Unified Messaging The network diagram shows two Business Communications Managers but additional base units can be added Figure 38 Multiple Business Communications Manager systems network diagram Head Office Warehouse System ME erie Systemi y Communications unicati telephone Kl nadel i Manager telephone PSTN fallback route Intranet Fal VolP trunk i i2050 Software Phone i2004 telephone remote i2004 To set up this system Ensure that the existing network can support the additional VoIP traffic Coordinate a Private dialing plan between all the systems On each Business Communications Manager 3 0 system e Set up outgoing call configuration for the VoIP gateway e Setup aremote gateway for the other Business Communications Managers or NetMeeting users IP Telephony Configuration Guide 118 Chapter6 Typical network applications using MCDN e Set telephones to receive incoming calls through target li
175. nks Table 16 Media parameters record Continued Field Value Description Jitter Buffer Voice Auto None Small Medium Large MaxVoiceJitterBuffer Select the size of jitter buffer you want to allow for your system Jitter Buffer Voice MaxVoiceJitterBuffer v Refer to Jitter Buffer on page 26 Outgoing call configuration This section explains how to set up your system to place calls through VoIP trunks The system at the other end of the call must be set up to receive VoIP calls For information about this refer to Incoming call configuration on page 89 Outgoing call configuration consists of the following steps e Putting VoIP lines into a line pool on page 76 e Configuring telephones to access the VoIP lines on page 78 e Configuring a remote gateway on page 78 e Optional Configuring PSTN fallback on page 80 Putting VoIP lines into a line pool Lines 001 to 060 are reserved for VoIP trunks However they can be used only if you have entered the appropriate keycodes to activate them When putting VoIP trunks into a line pool choose a line pool that is not used for any other type of line Once you have created a line pool you create an access code that the user dials on their telephone to access the line pool Note Set up an access code for the line pool only if you are NOT planning to use PSTN fallback If you intend to use PSTN fal
176. nology A link is busy for the duration of the packets Ping This utility is used to echo messages to a host over an IP network This allows you to find out if the other point is available Ping also can include statistics about how long it took from end to end which provides information about routing prioritization This refers to how a voice data packet is set up in the Business Communications Manager so that external routers recognize it as having a high priority thus shortening delay times and increasing the perception of voice quality over VoIP trunks published IP address The IP address that both the IP telephones and the Symbol NetVision telephones use to access the Business Communications Manager NetVision uses the H 323 RAS protocol QoS quality of service routing To minimize voice jitter over low bandwidth connections the Business Communications Manager programming assigns specific DiffServ Marking in the IPv4 header of the data packets sent from IP telephones During the packet journey through the network including any routers on that network the header specifies a level of priority service This is quality of service routing For QoS to be successful for IP telephony it must be end to end on the network IP Telephony Configuration Guide 170 Glossary silence compression silence suppression This is the utility that omits the data packets that occur when no one is talking during the IP trunk calls thus
177. ns Manager 3 0 system IP telephones are e None e Small 06 seconds e Medium 12 seconds e Large 18 seconds QoS routing To minimize voice jitter over low bandwidth connections the Business Communications Manager programming assigns specific DiffServ Marking in the IPv4 header of the data packets sent from IP telephones The DiffServ Code point DSCP is contained in the second byte of the IPv4 header DSCP is used by the router to determine how the packets will be separated for Per Hop Behavior PHB The DSCP is contained within the DiffServ field which was known as the ToS field in older versions The Business Communications Manager assigns Expedited Forwarding EF PHB for voice media packets and the Class Selector 5 CS5 PHB for voice signaling control packets On the Business Communications Manager these assignments cannot be adjusted The Business Communications Manager 3 0 system performs QOS routing but if one or more routers along the network route do not support QOS routing this can impact voice quality Business Communications Manager 3 0 system QoS can also be configured so that the system reverts to a circuit switched line if a suitable QoS cannot be guaranteed IP Telephony Configuration Guide 28 Chapter Introduction P0993474 03 29 Chapter 2 Prerequisites checklist Before you set up VoIP trunks or IP telephones on a Business Communications Manager complete the following checklists
178. nsfer mode SONET or wireless connection e Scalability A future proof flexible and safe solution combined with high reliability allows your company to focus on customer needs not network problems Nortel Networks internet telephony solutions offer hybrid environments that leverage existing investments in Meridian and Norstar systems IP Telephony Configuration Guide 20 Chapter 1 Introduction e Increased customer satisfaction Breakthrough e business applications help deliver the top flight customer service that leads to success By providing your customers with rapid access to sales and support personnel via telephone the Web and e mail your business can provide better customer service than ever before IP telephones and VoIP trunks This section describes two similar applications for IP telephony on the Business Communications Manager 3 0 system IP telephones and VoIP trunks These applications can be used separately or together as a network voice data solution e IP telephones on page 20 e VoIP trunks on page 20 IP telephones IP telephones offer the functionality of regular telephones but do not require a hardwire connection to the Business Communications Manager Instead they must be plugged into an IP network which is connected to the LAN or WAN card on the Business Communications Manager 3 0 Calls made from IP telephones through the Business Communications Manager can pass over VoIP trunks or acro
179. nt to forward the call Or if you select speed dial FEATURE 0 enter the speed dial code for the number you want the telephone to dial This feature allows you to assign your hardware feature keys to line and intercom applications and still access the Business Communications Manager call features without needing to remember a feature code Although the list is defaulted to the Services button you can assign the display list to one of the other hard feature keys The user can also assign it as a memory button using FEATURE 3 at a specific telephone Refer to the Programming Operations Guide for information about programming IP telephone memory buttons under User Preferences Note If you move the feature to another memory button the Services button no longer accesses the menu Using the Hot Desking feature You can transfer your IP telephony features temporarily from one IP telephone to another using the Hot Desking feature This feature is described in detail in the Telephony Feature Handbook You use FEATURE 999 to enter the feature To perform hot desking you are prompted for a password which is specified at the telephone before you can complete the task The Hot Desking password can be reset from the Unified Manager This allows users who forget their passwords to re enter hot desking and to reset their password Note This process also cancels hot desking for the telephone if the application is currently active
180. nta Clara Business Communications Manager using the following settings e Destination IP 47 62 84 1 This is the published IP address of the Santa Clara Business Communications Manager e QoS Monitor Enabled This must be enabled for PSTN fallback to function P0993474 03 Chapter 5 Configuring VoIP trunks 93 e Transmit Threshold 3 0 This is a Mean Opinion Score MOS value that ensures that the VoIP lines are used as long as the system can provide moderate quality e Receive Threshold 3 0 This is a MOS value that ensures that the VoIP lines are used as long as the system can provide moderate quality e Destination Digits 3 This number will also be used as part of the Destination code network and the 3 is included in the DN this number will be absorbed before dialout Note In this case because the systems are on a Coordinated Dialing Plan CDP gt 10 11 12 13 14 The installer sets up the VoIP schedule with these settings e Service Manual Overflow Y The installer ensures a route has been created to the line pool that accesses the local PSTN line including the external dialout The installer defines a new route called Route 003 and sets it to use line pool PRI A This is the line pool that contains the PRI fallback lines The installer defines a new route called Route 100 and sets it to use line pool O This is the line pool that contains the VoIP lines The installer creat
181. oIP Gateway monitoring function can take about three minutes to respond to marginal changes in the network condition Packet loss Packet loss is the percentage of packets that do not arrive at their destination Transmission equipment problems and high delay and congestion can cause packet loss In a voice conversation gaps in the conversation represent packet losses Some packet loss less than 596 can be acceptable without audible degradation in voice quality Packet delay Packet delay is the period between when a packet leaves and when a packet arrives at the destination The total packet delay time includes fixed and variable delay Variable delay is the more manageable delay while fixed delay depends on the network technology The distinct network routing of packets are the cause of variable delays To minimize packet delay and increase voice quality the gateway must be as close as possible to the network backbone WAN with a minimum number of hops P0993474 03 Quality of Service 163 Delay variation jitter The amount of variation in packet delay is otherwise known as delay variations or jitter Jitter affects the ability of the receiving gateway to assemble voice packets received at irregular intervals into a continuous voice stream Fallback to PSTN If the measured Mean Opinion Score MOS for all codecs is below the configured threshold for any monitored gateway the Fallback to PSTN activates This feature rerou
182. oM M rm 16 Presales Suppo em sa iaausqdQGc uE NAT ATRRTEYRQIXIAREEZSXad qFsemdak 16 EMEA Europe Middle East Africa coc cesses m m RR xm 16 CALA Caribbean amp Latin America 2 6 cca eee kx ak OR kon RR dn Rc AR a E 16 og CX C Rec o eT T E E EEE ET ETT T TE E 17 Chapter 1 Pete MTM PDCDEUTUITTTTTTITUT 19 IF telephones and VoIP Yunks iuis sni EuAd EAPe xc eh oye ees OY ee pride ass 20 IF dol DD S qacqcaemseessiqa emBeCerSc4amqexsqA e maDSwSqea RES REDE ROBES 20 gg ST eee es EEO REEE ESSEC ROEHL ERE Re R EER RR 20 Creating the IP telephony network Lass lasw d setas e ede be ranten a ede 2 Business Communications Manager 3 0 i i coeno a Rh yx Ry ewan RR 22 IER eerie veal vaio ahd Rd aca OIM C eer Rel aded een a Roda 23 or oar thes ee SR Ee dd eke ee ead eens ay Eke 23 VoIP trunks and analog digital telephones 0 cee eee eee 23 VoIP trunks and IP telephones sss oss ote re odo Re 24 Gatekeepers on the NGIWOIK 22a b usadao p ad rm a Rue a Ra RE Rope dd a Y Ra Rig 24 IP felWOIK chock a ceste ek ROR ERED arua edes ue dax dd dus Ed aac dris RR e d 24 SOUS Pas iscoita so dct est xc c Buy ct oc CER pu blau s Dat ah Rx UE akan A ea d 24 LA scene poet aepene wes diatouedss OTA deta peRew asst oueeeenacaGaas 25 Public Switched Telephone Network 0 00 e eee eee ee ees 25 Kay IP telephony ODDO DIS se dad e oue ied a a dede adhd candace ol desig die al dee c EC 25 d o P geld saisir aiea FO t a na u EES Be ee eee 26 SSA 15 24
183. om http lt Gatekeeper IP gt gk Before an endpoint registers with the CSE 1000 gatekeeper it must first be added to the gatekeeper configuration Before a registered endpoint may make calls it must have its numbering plan information assigned within the gatekeeper configuration Before any of these configuration changes become part of the gatekeeper active configuration they must be committed to the active database Configuration and activation information is described in the following sections IP Telephony Configuration Guide 110 Chapter5 Configuring VoIP trunks Business Communications Manager requirements Set the Business Communications Manager Local Gateway IP interface to the following Set Call Signaling Method to either GatekeeperResolved or GatekeeperRouted depending on your system requirements Set Gatekeeper IP to the IP address at which the CSE 1000 gatekeeper operates Set Alias Names to a single H 323 identifier that is unique across all endpoints registered with the gatekeeper For example NAME BCM OTTAW A This H 323 identifier must exactly match that in the CSE 1000 gatekeeper configuration This entry is case sensitive CSE 1000 configuration adding an H 323 endpoint In the Gatekeeper Admin tool perform the following A OO N Select GK standby DB admin Select H 323 Endpoints Select Add H 323 Endpoint Ensure the following fields are set Table 24 CSE 1000 H 323 endpoints
184. on 44 feature labels 59 firmware downloading 60 H 323 Terminals record 70 home based network 120 12050 Software Phone 64 installing 39 67 Invalid server address 48 Jitter buffer 43 jitter buffer 53 Keep DN Alive 63 keycode 68 network check list 29 New telephone 48 No ports left 48 prerequisites 37 Published IP address 46 register prompt 47 registering 40 Registration disabled 48 relocating 63 restart to configure 45 review configuration information 48 router IP 46 server parameters 46 Set IP viewing 48 settings 53 slow connection 48 speech paths 49 staggered download 61 Troubleshooting 48 troubleshooting prompts 48 Unified Manager configuration 53 updating H 323 terminals record 71 VLAN service 37 VLAN settings 47 148 IP telephony asymmetrical media channel negotiation 145 Benefits 19 concepts 25 engineering link capacity 130 insufficient link capacity 130 Introduction 19 network checklist 29 network loading 128 network DHCP 49 networks 21 ongoing monitoring 134 setting QoS 153 WAN link resources 121 IP Terminal status 53 IP terminal status features list 55 IP trunks media resources 31 network prerequisites 29 IP TTL Traceroute 141 IP wireless keycode 68 IPWIs NetVision mode 71 J jitter 163 Jitter buffer adjust size 133 defined 26 IP telephones 43 Unified Manager settings 53 VoIP trunks 76 K Keep DN alive 63 keycodes IP telephones 39 NetVision 68 prerequisite list 33 VoIP trun
185. ons Each H 323 or VoIP Realtime Transfer Protocol RTP flow uses two ports for each direction The total number of UDP port numbers to be reserved depends on how many concurrent RTP flows are expected to cross a router interface In general e Backbone routers reserve more ports than edge routers e The port ranges on edge routers are a subset of the backbone router port ranges e Include port number UDP 5000 in the reserved port ranges for the QoS monitor e The port ranges reserved in a Business Communications Manager 3 0 system are also reserved by the remote router e You must reserve two ports for each voice call you expect to carry over the WAN link e You can reserve multiple discontinuous ranges Business Communications Manager 3 0 requires that each range meet the following conditions Each range must start with an even number Each range must end with an odd number You cannot have a total of more than 256 ports reserved Using a gatekeeper This section describes the use of a gatekeeper for your VoIP trunks e Modifying the Local Gateway Settings on page 104 e Using Radvision ECS 2 1 0 1 as the gatekeeper on page 108 e Using CSE 1000 as a gatekeeper on page 109 e Gatekeeper call scenarios on page 113 The Business Communications Manager supports the use of an ITU H323 gatekeeper A gatekeeper is a third party software application residing somewhere on the network which provides servi
186. oper phony Services Feature 8 984 stem DNs es Label 8 Cfwd vmail ps striction filters Feature 9 985 I routing 1eduled services stem speed dial neral settings Feature settings Nortel IP terminals 9 Feature labels Labels 1 6 Label 3 VMail DN Feature 10 986 Label 10 xfer vmail Feature 11 987 Label 11 Mail intr ees perennem emend Labels 13 18 Pabst amma IP Telephony Configuration Guide 60 Chapter3 Installing IP telephones 3 If you have an existing list or you do not want to change any defaults go to the first free label set 4 Inthe Feature label number field enter the dialing code for the feature you want to relabel Example enter 3 for conference call 5 In the Label label number field enter the new label you want the telephones to display Example The current label for feature code 3 is Conference you could change it to Conf Call 6 Click anywhere outside the field to save the changes The system automatically updates any 12002 12004 or 12050 IP telephones that have a button appearance for the feature Some features like Page and System Wide Call Appearances SWCA have several variations of feature invocation that you may want to customize for the users Paging can be F60 F61x F62 and F63x System wide Call Appearance SWCA has 16 codes 521 to 536 The following table shows examples of changing labels for page codes and SWCA codes Table 14 Relabelling examp
187. ore upgrading a link check both routers connected to the link for the upgrade and ensure correct router configuration guidelines Change the link from satellite to terrestrial to reduce the link delay by approximately 100 to 300 ms Put into operation a priority queueing rule Identify the links with the highest use and the slowest traffic Estimate the link delay of these links using Traceroute Contact your service provider for help with improving your QoS Reducing hop count To reduce end to end delay reduce hop count especially on hops that move across WAN links Some of the ways to reduce hop count include Improve meshing Add links to help improve routing adding a link from router1 to router4 instead of having the call routed from router to router2 to router3 to router4 reducing the hop count by two Router reduction Join co located gateways on one larger and more powerful router P0993474 03 Efficient Networking 133 Adjust the jitter buffer size The parameters for the voice jitter buffer directly affect the end to end delay and audio quality IP telephony dynamically adjusts the size of the jitter buffer to adjust for jitter in the network The network administrator sets the starting point for the jitter buffer Lower the jitter buffer to decrease one way delay and provide less waiting time for late packets Late packets that are lost are replaced with silence decreasing quality Increase the size of the jitt
188. ore you continue with this section For information about creating a VoIP line pool see Putting VoIP lines into a line pool on page 76 You can create a PSTN line pool in the same manner if such a pool does not already exist Note If you already have routes for your PSTN or VoIP line pools configured you do not need to configure new routes unless you cannot match the dialed digits For instance you probably already have a PSTN route that uses 9 to access local PSTN numbers P0993474 03 Chapter 5 Configuring VoIP trunks 83 Follow these steps to configure the PSTN and VoIP routes Open Services Telephony Services Call Routing and click on Routes 2 Enter the route numbers for the PSTN and VolP lines PSTN to other system a Click the Add button The Add Routes dialog appears Figure 23 Add route dialog Add Routes b c Type a number between 001 and 999 to define the PSTN route to the other system Only numbers not otherwise assigned will be allowed by the system Click Save PSTN to local PSTN lines a Click the Add button b Inthe Add routes dialog Route field type a number between 001 and 999 to define the PSTN route to your local PSTN Only numbers not otherwise assigned will be allowed by the system C Click Save VoIP a Click Add button b Inthe Add routes dialog Route field type a number between 001 and 999 to define the VoIP route C Click Save 3 Assign the line
189. ork on page 24 e TP network on page 24 e Public Switched Telephone Network on page 25 The following figure shows components of a Business Communications Manager 3 0 network configuration Note that the two Business Communications Manager systems are connected both through a PSTN connection and through a WAN connection The WAN connection uses VoIP trunks If the PSTN connections use dedicated ISDN lines the two systems have backup private networks to each other Both Business Communications Manager systems use VoIP trunks through a common WAN to connect to the Meridian M1 ITG system IP Telephony Configuration Guide 22 Chapter1 Introduction Figure 1 Network diagram Business Communications Manager A Router NetVision H 323 device A 12050 telephone A Gatekeeper mE Business Communications Manager B M1 ITG IP tel fn d elephone ia Meridian set A Business Communications Manager 3 0 The Business Communications Manager 3 0 is a key building block in creating your network It interoperates with many devices including the Meridian 1 system and H 323 devices In the diagram shown in Figure on page 22 the Business Communications Manager 3 0 system is connected to devices through multiple IP networks as well as through the PSTN Multiple Business Communications Manager 3 0 systems also can be linked together on a network of VoIP trunks and or dedicat
190. orm to internal quality of service standards and that QoS objectives are always met P0993474 03 135 Appendix B Silence compression This section describes using silence compression on half duplex and full duplex links e Silence Compression on Half Duplex Links on page 136 e Silence compression on Full Duplex Links on page 138 e Comfort noise on page 140 Silence compression reduces bandwidth requirements by as much as 50 per cent This section explains how silence compression functions on a Business Communications Manager network For information about enabling silence compression in VoIP gateways refer to Configuring media parameters on page 74 G 723 1 and G 729 Annex B support Silence compression A key to VoIP Gateways in business applications is reducing WAN bandwidth use Beyond speech compression the best bandwidth reducing technology is silence compression also known as Voice Activity Detection VAD Silence compression technology identifies the periods of silence in a conversation and stops sending IP speech packets during those periods Telco studies show that in a typical telephone conversation only about 36 to 40 of a full duplex conversation is active When one person talks the other listens This is half duplex There are important periods of silence during speaker pauses between words and phrases By applying silence compression average bandwidth use is reduced by the same amount This r
191. perability 145 network prerequisites 29 signaling method 104 Gatekeeper IP Local Gateway 106 GateKeeperResolved 106 GateKeeperRouted 106 gateway Business Communications Manager QoS support 161 connecting to intranet 131 destination digits 85 H 323 specifications 143 IP telephones 46 monitoring QoS 154 network prerequisites 29 progress tones 146 remote configuring 78 Gateway Protocol 78 Gateway Protocol Local Gateway 107 Gateway Type 78 Global IP see Published IP address 35 GWProtocol 107 IP Telephony Configuration Guide 174 Index H H 323 gateway specifications 143 non linear processing 127 Trunks record jitter buffers 76 H 323 devices NetMeeting 143 NetVision 67 H 323 endpoints 104 H 323 terminals record deleting handset record 72 NetVision 70 updating 71 H 323 Trunks record 74 activating QoS monitor 87 enabling PSTN fallback 81 remote gateway 78 H323ldentifier 106 half duplex links bandwidth requirements 124 silence compression example 136 silence suppression 124 handset changing name 72 deleting record 72 home based users 120 hop count reducing 132 hot desking change password 57 Hunt group target line to DN 89 12002 connecting 96 server parameters 46 12004 connecting 96 feature labels 59 keep DN alive 63 server parameters 46 12050 Software Phone configuring 64 keep DN alive 63 server parameters 46 IEEE Address H 323 terminals list also see ESS ID 71 inappropriate
192. r more than one endpoint The gatekeeper will select the endpoint with the lowest EntryCost value 7 Click Create Committing Gatekeeper Configuration Changes Gatekeeper changes occur in the standby database For these settings to be used by the active gatekeeper you must commit them to the active database from the Gatekeeper Admin tool as describe below 1 Select GK Standby DB Admin 2 Select Database Actions 3 Select Single Step Commit and Crossover Configuring Codec Compatibility The default codec settings for a CSE1000 are not compatible with those used by a Business Communications Manager system In order to successfully make IP trunk calls between a Business Communications Manager and the CSE 1000 the codec configuration on both the Business Communications Manager and the CSE 1000 must coincide as shown in the table below As well any configured codecs on the CSE 1000 must have their payload size set to 30 ms Caution The CSE 1000 can only register five codecs at once This can include G 711 mu law G 711 a law T 38 G 711CC and either G 729A G729AB or G 723 1 It is important to that you disable the unused codecs This ensures that the required codecs get registered with the DSP Failure to disable unused codecs could result in the wrong codecs being registered with the DSP which would create call failures IP Telephony Configuration Guide 112 Chapter5 Configuring VoIP trunks Table 26 CSE1
193. ranet delay characteristics get Ping measurements over a period of at least a week Measuring end to end packet loss The Ping program also reports if the packet made its round trip correctly Use the same Ping host setup to measure end to end errors Use the same packet size Sampling error rate require taking multiple Ping samples at least 30 An accurate error distribution requires data collection over a greater period of time The error rate statistic from multiple Ping samples is the packet loss rate Recording routes As part of the network evaluation record routing information for all source destination pairs Use the Traceroute tool to record routing information A sample of the output of the Traceroute tool follows C WINDOWSstracert 10 10 10 15 Tracing route to 10 10 10 15 over a maximum of 30 hops 13 ms 1 ms 10 ms tftzrafl ca nortel com 10 10 10 1 2 1 ms 1 ms 1 ms 10 10 10 57 3 7 ms 2 ms 3 ms tcarrbf0 ca nortel com 10 10 10 2 4 8 ms 7 ms 5 ms bcarha56 ca nortel com 10 10 10 15 Trace complete The Traceroute program checks if routing in the intranet is symmetric for each source destination pairs Also the Traceroute program identifies the intranet links used to carry voice traffic For example if Traceroute of four site pairs gets the results shown in the following table you can calculate the load of voice traffic per link as shown in the second table Table 39 Site pairs and routes Site pair Intr
194. rea Local Exchange Carrier Mean Opinion Score NetVision Phone Administrator Pulse Code Modulation Packet InterNet Groper Power inline patch panel Point to Point Protocol Primary Rate Interface Public Switched Telephone Network Quality of Service Registration Admissions and Status Real time Transfer Protocol Simple Network Management Protocol Transmission Control Protocol User Datagram Protocol UNISTIM Terminal Proxy Server Voice over Internet Protocol Voice Activity Detection Virtual LAN Wide Area Network IP Telephony Configuration Guide 16 Preface Related publications Documents referenced in the Business Communications Manager 3 0 IP Telephony Configuration Guide include Installation and Maintenance Guide Software Keycode Installation Guide e Programming Operations Guide Telephony Feature Handbook e 12004 12005 12050 Software Phone user cards How to get help USA and Canada Authorized Distributors ITAS Technical Support Telephone 1 800 4NORTEL 1 800 466 7835 If you already have a PIN Code you can enter Express Routing Code ERC 196 If you do not yet have a PIN Code or for general questions and first line support you can enter ERC 338 Website http www nortelnetworks com support Presales Support CSAN Telephone 1 800 4NORTEL 1 800 466 7835 Use Express Routing Code ERC 1063 EMEA Europe Middle East Africa Technical Support CTAS Telephone 00
195. records refer to the Business Communications Manager Programming Operations Guide To activate the VoIP schedule 1 Dial FEATURE 873 from the control set for the VoIP trunk The phone prompts you for a password Type the password Press OK The first schedule appears Scroll down the list until VoIP is selected Press OK The VoIP schedule stays active even after a system reboot and can only be deactivated manually To deactivate the VoIP schedule 1 Dial FEATURE 873 The phone prompts you for a password 2 Type the password 3 Press OK The system returns to the Normal schedule Turning on QoS monitor For fallback to function the QoS monitor must be enabled 1 In Unified Manager open Services IP Telephony H 323 Trunks and click on Remote Gateways The Remote Gateway screen appears Select the Remote Gateway listing for which you want to enable QoS Monitoring On the top menu click Configuration then click Modify Entry The Remote Gateway dialog appears 4 For the QoS Monitor field select Enabled IP Telephony Configuration Guide 88 Chapter5 Configuring VoIP trunks Figure 26 QoS Monitor field on the Remote Gateway screen Disabled x QoS Monitor Disabled smit Threshold 5 Set the Transmit Threshold and Receive Threshold to a value between 0 and 5 Figure 27 Threshold fields on the Remote Gateway screen Receive Threshold o This marks the level of quality that the ga
196. reducing the bandwidth load required for IP calls Symbol NetVision handsets These IP telephones connect to the system through wireless access points connected to the same network to which the Business Communication Manager is connected target lines These are internal channels on the Business Communications Manager that allow you to direct incoming calls to specific telephones call groups Hunt groups or system devices System telephones require target lines if they have not already been configured when receiving VoIP trunk calls so the call knows where to go terminal Device capable of sending or receiving data over a data communications channel throughput Indicator of data handling ability Measures data processed as output by a computer communications device link or system topology Logical or physical arrangement of nodes or stations Traceroute Traceroute uses the IP TTL time to live field to determine router hops to a specific IP address UNISTIM Terminal Proxy Server UTPS This is a Nortel designed protocol for IP telephony applications The 12004 and 12002 for instance use this protocol to communicate with the Business Communications Manager voice compression Method of reducing bandwidth by reducing the number of bits required to transmit voice Voice over IP VoIP trunks VoIP trunks are virtual telephone lines that the Business Communications Manager uses instead of wired lines to transfer
197. s On the H 323 Terminal List screen highlight the terminal you want to change At the top of the page click on Configuration menu and select Update Entry The H 323 Terminal List dialog appears Enter a new password Click Save IP Telephony Configuration Guide 72 Chapter4 Installing NetVision telephones Changing a handset Name The Name is the primary point of recognition for the Business Communications Manager to identify a handset If you need to change the name of an assigned handset 1 Delete the existing record Refer to Deleting a NetVision telephone from the system on page T4 2 Enter a new record with the new name You can assign the existing DN to the new record 3 For security purposes you should assign a new Password Changing the DN record of a handset If you need to change the DN for a handset use the Unified Manager Services Telephony Services General Change DN The change will automatically be reflected in the H 323 Terminals record for the handset When you use the Change DN feature the DN settings are transferred to the new DN and the system features remain active on the new DN Warning Deleting an H 323 Terminals record will remove the DN from the Active DNs N list This means that system features such as Call Forward No Answer will also become inactive Deleting a NetVision telephone from the system If you want to stop a terminal from having access to the Business
198. s a Normal and VoIP schedule that includes the route you created to the local PSTN Note If you already have a line pool access code defined as 9 you will need to delete this record before you create the destination code Follow these steps to create destination codes for your fallback route 1 Open Services Telephony Services Call Routing and highlight Destination Codes 2 Click Add The Add Destination codes dialog appears Add Destination codes P0993474 03 Chapter 5 Configuring VoIP trunks 85 3 Type a one or more digits for this destination code Note For example if it is available you might want to use the same number s that you used for the destination code of the gateway If you have multiple gateways you could use a unique first number followed by the destination digits to provide some consistency such as 82 83 84 85 to reach gateways with destinations digits of 2 3 4 and 5 Click Save to close the dialog Click on the destination code heading for the destination code you just created Click on the key beside Schedules and highlight VoIP The VoIP schedule appears as shown in the next figure Figure 24 VoIP schedule i Destination codes Q 5A 9 SA VoIP Del All r5A VoIP First Route Route 500 v Schedules Normal Absorbed length x Night Evening Second Route None x Lunch Absorbed length All vj Sched 5 Th
199. s of configuring handsets to the Business Communications Manager e You must have an H 323 record configured before you configure the handsets with the Nortel NVPA e Each telephone that you configure will use one IP client assignment so ensure that you added enough keycodes to accommodate both your IP telephones and your NetVision telephones e Ifyou do not specify a DN in the H 323 record one will automatically be assigned to the handset If you specified a DN record it will appear under the Active DNs heading once the handset connects to the system If you want to specify a range of DNs you can use the Add Users Wizard This wizard is explained in the Business Communications Manager 3 0 Programming Operations Guide IP Telephony Configuration Guide 70 Chapter4 Installing NetVision telephones e You need to set up the DN record to determine what lines the handset can access and how it will behave on the system e The Name you specify in the H 323 record must match the User Name you specify in the Nortel NVPA tool otherwise the handset will not be allowed to connect to the Business Communications Manager If you need to change the H 323 Terminals record refer to Updating the H 323 terminals record on page 71 and Deleting a NetVision telephone from the system on page 72 If you require information about changing the DN records refer to the Business Communications Manager 3 0 Programming Operations Guide for deta
200. sed at 48 during the peak hour with a planning limit of 85 has an available capacity of approximately 568 kbit s Network engineering This section describes some network engineering criteria that you need to consider for your system e Bandwidth requirements on half duplex links on page 124 Bandwidth requirements on full duplex links on page 125 LAN engineering examples on page 126 e WAN engineering on page 126 e QoS Monitoring Bandwidth Requirement on page 127 Engineer the network for worst case numbers to indicate the spare bandwidth a LAN must have to handle peak traffic It is important to plan so that the LAN WAN can handle the IP telephony traffic using the defined codec without delay or packet loss The installer or administrator must select one configuration and then set up the LAN WAN so there is more bandwidth than the IP telephony output The following table provides bandwidth characteristics for the transmission of voice over IP for various link types given codec type and payload sizes The bandwidths provided in this table explain the continuous transmission of a unidirectional media stream Table 28 VoIP Transmission Characteristics for unidirectional continuous media stream Codec Type Payload Size IP Packet Ethernet B W PPP B W FR B W ms Bytes Bytes kbit s kbit s kbit s G 711 10 80 120 116 8 97 6 103 2 or 20 160 200 90 4 80 8 83 6 30 240 280 81 6 75 2 77 1
201. ses 108 Gatekeeper support for interoperability llle 109 Using CSE 1000 as a gatekeeper eee 109 Business Communications Manager requirements sues 110 CSE 1000 configuration adding an H 323 endpoint 000 110 Setting the H 323 Endpoint Dialing Plan iiussscusiiuu RR ER RR REEL 110 Committing Gatekeeper Configuration Changes 0 0c eee eee 111 Configuring Codec Compatibility 0 00 eee ee 111 alee er cal SCORE Lois Cal eb x ndash dba abb R E Du E Cakes 113 Chapter 6 Typical network applications using MCDN llle 115 Setting up MCDN over VoIP with fallback leeen BS 115 MCDN functionality on fallback PRI lines 0 eee eee 116 Networking multiple Business Communications Managers liliis 117 Multi location chain with call Genial isssocse a st RE e RR RERERERPTCRER P EY ERR 119 Business Communications Manager to IP telephones lseelseess 120 Setting up a remote based IP telephone 00 0 00 cece eee eens 120 Appendix A Efficient Networking iis sas RRARRARRKEARARAERRKAAERARARAXRRARRAEAARARRAA 121 Determining the bandwidth requirements l lselseeeleeee eee 121 Determining WAN link resources eessseeeelee eee 121 LAKO TIA O NND rm 121 Lin nesgcuip nls MP TCI p AaRON SR a 122 Bandwidth requirements on half duplex links 0 00 eee 124 Bandwidt
202. specific information about configuring the gatekeeper for each application Fallback to Circuit Switched Enabled All Your choice determines how the system will handle calls if Enabled TDM only the IP network cannot be used Disabled Enabled All All calls will be rerouted over specified TDM SCNFallback trunks lines Fallback to Circuit Switched 5 CNFallback v Enabled All Enabled TDM only Gatekeeper IP Disabled Call Signaling e Enabled TDM only All voice calls will be rerouted over specified TDM trunks lines Disabled Calls will not be rerouted IP Telephony Configuration Guide 106 Chapter5 Configuring VoIP trunks Table 19 Local Gateway IP interface fields Continued Field Value Description Call Signaling Direct Direct call signaling information is passed directly GateKeeperRouted between endpoints The remote gateway table in the Unified Manager defines a destination code digits for GateKeeperResolved each remote system to direct the calls for that system to CallSignaling route In each system the Nortel IP Terminals and H 323 Terminals records map IP addresses to specific telephones e Gatekeeper Resolved all call signaling occurs directly between H 323 endpoints This means that the gatekeeper resolves the phone numbers into IP addresses but the gatekeeper is not involved in call signaling e Gatekeeper Routed uses a gatekeeper for call setup and contro
203. ss a Public Switched Telephone Network PSTN Nortel Networks provides two types of IP telephones The IP telephones are wired to the IP network using Ethernet in the case of the 12002 and the 12004 or are accessed through your desktop or lap top computer as in the case of the Nortel Networks 12050 Software Phone Emobility voice can be provided using Symbol NetVision or NetVision Data telephones which connect through an access point wired to an IP network configured on the LAN NetVision telephones use an extended version of the H 323 protocol to connect to the system VoIP trunks VoIP trunks allow voice signals to travel across IP networks A gateway within the Business Communications Manager 3 0 converts the voice signal into IP packets which are then transmitted through the IP network to a gateway on the remote system The device at the other end reassembles the packets into a voice signal BCM ITG Meridian and NetMeeting are devices that can use the H 323 protocol trunks which the 3 0 Business Communications Manager system supports P0993474 03 Chapter 1 Introduction 21 Creating the IP telephony network This section explains the components of the Business Communications Manager 3 0 system and the devices it interoperates to create a network This section includes information about Business Communications Manager 3 0 on page 22 e MI ITG on page 23 e Telephones on page 23 e Gatekeepers on the netw
204. ssessment 31 P0993474 03 Index 173 E E 164 106 echo cancellation 143 echoreply 141 efficient networking 121 Enable TTL 108 end to end delay 131 155 end to end DTMF signaling 143 Endpoint Type Radvision 109 end to end packet loss measuring 156 errors gathering statistics 158 network analysis 131 ethernet B W 122 124 125 ethernet connection IP telephones 44 external 85 F fallback activating VoIP schedule 87 configuring for PSTN 80 destination codes 84 enabling 81 MCDN 115 MCDN networking 116 Mean Opinion Score 163 MOS for codecs 163 scheduling 82 using PRI line 91 VoIP line pools 76 Fallback to Circuit Switched Local Gateway 105 fastStart 145 FEATURE hot desking 999 58 features i2004 labels 59 features list 55 services key 900 56 filtering criteria 147 ranges 147 firewall IP configuration note 49 firewalls configuring 101 network prerequisites 30 ports 101 firmware downloading to IP telephones 60 Force Direct for Service Calls Radivision 108 force download 60 Force Online Status Radvision 109 FR B W 122 124 125 Frame Relay 122 full duplex link bandwidth requirements 125 silence compression examples 138 silence suppression 125 VoIP load 129 WAN engineering 126 G G 711 122 124 125 723 1 122 124 125 G 729 122 124 125 Gatekeeper interoperability support 109 Radivision ECS 2 1 0 1 108 gatekeeper 103 call scenarios 113 defined 24 intero
205. st reboot your Business Communications Manager after you enter VoIP keycodes to activate trunking If you want to use the MCDN features on the VoIP trunks you will need an MCDN keycode However if you have already deployed MCDN for your SL 1 PRI lines you do not require an additional keycode Published IP address You will require the public IP address to set up the gateways for VoIP trunks Refer to Defining published IP address on page 35 for details Configuring media parameters You can use the screen described in this section to determine the order the system will select codecs for your IP terminals the silence suppression settings and the jitter buffers In Unified Manager click on the keys beside Services IP Telephony Click on H 323 trunks Click on the Media Parameters tab The Media Parameters dialog appears Figure 17 Media parameters Local Gateway IP Interface Media Parameters Media Parameters 1st Preferred Codec 8 723 v 2nd Preferred Codec G 723 v 3rd Preferred Codec G 711 uLaw gt 4th Preferred Codec G 711 aLaw v Silence Compression Disabled Jitter Buffer Voice auro vj P0993474 03 Chapter 5 Configuring VoIP trunks 75 4 Use the information in the table below to set up the media parameters for your system Table 16 Media parameters record SilenceCompression Field Value Description 1st Preferred Codec None Se
206. system Otherwise choose Disabled For information about enabling QoS see Turning on QoS monitor on page 87 Transmit Threshold read only Receive Threshold read only Gateway Type BCM3 0 Choose the type of system that is accessed through the remote BCM2 5 gateway BCM2 0 BCMG 0 Business Communications Managers running 3 0 software ITG BCM2 5 Business Communications Managers running 2 5 or 2 5 CSE 1000 FP1 or FP1 Maintenance Release software CS 3000 BCM 2 0 Business Communications Managers running 2 0 software or Enterprise Edge systems running 2 0 x software IMS ITG M1 Internet Telephony Gateway CSE 1000 CS3000 IMS CS3000 is the previous version of IMS If your gateway is set to BCMX X and the other system is upgraded to 3 0 your system will automatically update this listing to BCM3 0 when the other system is contacted after the upgrade If this does not occur your original configuration may not be correct and you will have to set the change manually IP Telephony Configuration Guide 80 Chapter5 Configuring VoIP trunks Table 17 Remote gateway record Continued could be the same as the destination code for the route to this system Field Value Description Gateway Protocol None Select the gateway protocol that the trunk expects to use SL 1 None No special features CSE SL 1 MCDN protocol for gateways that provide MCDN over VoIP service CSE Use this setting when using a CSE 100
207. tations that can communicate as if they were on the same LAN They also facilitate easier administration of move add and change in members of these groups e Traffic between VLANs is restricted Bridges forward unicast multicast and broadcast traffic only on LAN segments that serve the VLAN to which the traffic belongs e For IP telephony VLANs provide a useful technique to separate and prioritize the telephony traffic for L2 switches e VLAN also provide a shield from malicious traffic that may be targeted at the IP phone in order to steal or disrupt service e Reuse IP address in different VLANs e As far as possible VLANs maintain compatibility with existing bridges and end stations e Ifall bridge ports are configured to transmit and receive untagged frames bridges will work in plug and play ISO IEC 15802 3 mode End stations will be able to communicate throughout the Bridged LAN Choosing DHCP for VLAN If you use a DHCP server remote to your Business Communications Manager you must enter any VLAN IDs manually on 12004 telephones By using the Business Communications Manager DHCP server you can configure DHCP to auto assign a VLAN ID to each IP telephone that registers With this configuration you can also choose to manually enter VLAN IDs if you choose The Business Communications Manager DHCP server becomes the default VLAN that everyone can reach The server provides the network configuration information in the default VL
208. telephony traffic from bursty LAN traffic and makes easier the end to end Quality of Service engineering for packet delay jitter and packet loss Further network analysis This section describes how to examine the sources of delay and error in the intranet It also discusses several methods for reducing one way delay and packet loss The key methods are e Components of delay on page 131 e Reduce link delay on page 132 e Reducing hop count on page 132 e Routing issues on page 134 Components of delay End to end delay is the result of many delay components The major components of delay are Propagation delay Propagation delay is the result of the distance and the medium of links moved across Within a country the one way propagation delay over terrestrial lines is under 18 ms Within the U S the propagation delay from coast to coast is under 40 ms To estimate the propagation delay of long haul and trans oceanic circuits use the rule of thumb of 1 ms per 100 terrestrial miles If a circuit goes through a satellite system estimate each hop between earth stations adds 260 ms to the propagation delay e Serialization delay The serialization delay is the time it takes to transmit the voice packet one bit at a time over a WAN link The serialization delay depends on the voice packet size and the link bandwidth and is the result of the following formula serialization delay in ms 8 IP packet size in bytes link b
209. ter about four seconds the top light flashes and NORTEL NETWORKS appears on the screen 2 When the greeting appears immediately and quickly press the four display keys one at a time from left to right These keys are located directly under the display These keys must be pressed one after the other within 1 5 seconds or the telephone will not go into configuration mode f Manual Cfg DHCP 0 no 1 yes appears on the screen you successfully accessed the configuration mode f any other message appears disconnect then reconnect the power and try to access the configuration mode again 3 Enterthe network parameters as prompted As each parameter prompt appears use the keypad to define values IP Telephony Configuration Guide 46 Chapter 3 Installing IP telephones Use the key to enter the period in the IP addresses Press OK to move forward The following table describes the values for each display parameter Table 10 IP telephone server configurations Field Value Description DHCP O or 1 Enter 0 if your network is not using a DHCP server to dispense IP addresses Partial DHCP Enter 1 if your network does use a DHCP server If you choose to use a DHCP server rather than allocating static IP addresses for the IP telephones skip the remainder of this section For information about setting up DHCP server information for the IP telephones see Configuring DHCP on page 49 SET IP
210. ter option and Subnet Mask option to configure along with IP address and lease time Full DHCP support in the 12004 terminal requires sending a Class Identifier option with each DHCP Discovery and Request message Additionally the 12004 checks for either a vendor specific option message with a specific unique to Nortel 12004 encapsulated sub type OR a site specific DHCP option In either case a Nortel 12004 specific option must be returned by the 12004 aware DHCP server in all Offer and Ack messages The 12004 will use the information P0993474 03 Chapter 3 Installing IP telephones 51 returned in this option to configure itself for proper operation This includes binding a new IP address netmask and gateway for local IP stack as well as configuring Server 1 minimum and optionally Server 2 By default Server 1 is always assumed to be the primary server after a DHCP session The 12004 will not accept any Offers Acks if they do not contain a Router option 12004 needs a default router to function AND Subnet Mask option AND an S1 Server Address and Port The i20XX sets require the scope value 128 to be configured on the DHCP server as follows Format Nortel i2004 A iii jjj kkk lll ppppp aaa rrr iii jjj kkk 111 p ppp aaa rrr where Nortel i2004 A uniquely identifies this as the Nortel option Additionally the A signifies this version of this specification Future enhancements could use B for example ASCII
211. tes calls to different trunks such as the Public Switched Telephone Network PSTN until the network QoS improves When the QoS meets or exceeds the threshold calls route over the IP network Fallback can be caused by any of the following reasons bad network conditions remote gateway is out of service e nonetwork connection e not enough DSP resources available The fallback feature can be in the Local Gateway Configuration With the fallback feature disabled calls move across the IP telephony trunks no matter what level of Quality of Service The fallback feature is active only at call setup A call in progress does not fall back if the quality degrades Calls fallback if there is no response from the destination an incorrectly configured remote gateway table or if there are not enough DSP resources available to handle the new call IP Telephony Configuration Guide 164 Quality of Service P0993474 03 165 Glossary access point 802 11b This is a piece of hardware using either IEEE 802 11 1 or 2 M bits sec Frequency Hopping Spread Spectrum or IEEE 802 11B 11 M bits sec Direct Sequence Spread Spectrum technology that connects to the internet and acts as a wireless gateway for devices to connect to the internet In the context of the Business Communications Manager this is the device that the NetVision handset uses to connect to the LAN to which the Business Communications Manager is connected bac
212. teway Codec G 711 v Reserved Media Gateway Codec should be set to whatever is the most commonly used codec for Media Gateways It determines the amount of codec resources reserved for each Media Gateway Reserving resources speeds up establishment of connections For example if most calls through a Media Gateway use the G 711 codec set this to G 711 If most calls use G 729 set this to G 729 Note that the higher the setting G 723 gt G 729 gt G 711 the more resources are set aside for Media Gateways This may result in calls failing to go through because of lack of available resources For a more detailed descriptions of the media gateway or other information about the media services card MSC settings for the Business Communications System refer to the Programming Operations Guide MSC section Keycodes All elements of VoIP trunks and IP telephony are locked by the Business Communications Manager keycode system You can purchase keycodes for the amount of access you want for your system Additional keycodes can be added later providing there are adequate resources to handle them Table 6 Keycodes Prerequisites Yes No keycodes 5 a Complete this question only if you are using VoIP trunks Do you have enough VoIP 5 b Complete this question only if you are using IP telephones Do you have enough IP client keycodes Note IP clients and IP telephones are a 1 1 ratio Include any NetVision tel
213. teway must be able to support before transmitting a call In most cases the transmit threshold and receive threshold should be the same On a line where communications in one direction are more important than in the other direction you can set up asymmetrical thresholds Warning QoS monitor must be turned on at both endpoints For information about using the QoS monitor refer to Quality of Service Monitor on page 100 PSTN fallback metrics To view the metrics associated with VoIP calls that fallback to the PSTN network 1 Choose Diagnostics Service Metrics Telephony Services and click the PSTN fallback metrics heading The Last reset time Fallback requests and Fallback failures values appear Figure 28 Fallback Metrics fields f PSTN fallback metrics Last reset time 20000101010000 Fallback requests 0 Fallback failures o To reset the metric log on the Configuration menu click Clear data and time P0993474 03 Chapter 5 Configuring VoIP trunks 89 Incoming call configuration To receive an incoming call directly to the telephone from a VoIP network you need to ensure that the telephone is mapped to a target line For information about setting up your Business Communications Manager to place outgoing VoIP calls see Outgoing call configuration on page 76 Assign a target line to the DN A target line routes incoming calls to specific telephones
214. the call is received over the VoIP trunk System B recognizes the code as its own and uses a local target line to route the call to the correct telephone Figure 30 Calling into a remote node from the PSTN Target line XXX recognizes 2244 DN 2244 DN 3322 DN 2244 assigned with target line XXX A Dian XX2 2244 Santa Clara Remote gateway set up to Ottawa CDP system code 2 Dialout 2244 Santa Clara Gateway 2 IP network n Gateway destination digit 2 dedicated VoIP trunk Gateway 3 Route 022 VoIP na PA DN type Private Destination code 2 using route 022 Absorb length 0 Ensure VoIP trunk is set up with remote filters Remote access over VoIP trunks You cannot program DISA or auto answer for voice over IP VoIP trunks therefore your system cannot be accessed from an external location over a VoIP trunk The exception to this is if the call comes into a tandemed system system A from a PSTN and the call is then sent out across a VoIP trunk to system B In this case system A is controlling remote access before transferring the call to system B through private routing Therefore all call features except Page are available to the caller depending on what the remote access package for the COS password allows IP Telephony Configuration Guide 98 Chapter5 Configuring VoIP trunks Configuring Net Meeting c
215. ti office telephony administration Telephones The Business Communications Manager 3 0 system can communicate using digital telephones M7000 T7000 T7100 M7100 M7100N T7208 M7208 M7208N T7316 M7310 M7310N M7324 and M7324N cordless telephones Companion DECT T7406 IP telephones and applications 12002 12004 Nortel Networks 12050 Software Phone and IP wireless telephones NetVision and NetVision Data telephones With this much flexibility the Business Communications Manager can provide the type of service you require to be most productive in your business VoIP trunks and analog digital telephones While analog and digital telephones cannot be connected to the Business Communications Manager 3 0 system with an IP connection they can make and receive calls to and from other systems through VoIP trunks Calls received through the VoIP trunks to system telephones are received through the LAN or WAN card and are translated within the Business Communications Manager to voice channels IP Telephony Configuration Guide 24 Chapter1 Introduction VoIP trunks and IP telephones The IP telephones connect to the Business Communications Manager across an IP network through either a LAN or a WAN From the Business Communications Manager connection they can then use standard lines or VoIP trunks to communicate to other telephones on other public or private networks The Business Communications Manager also supports H 323 and H
216. tics 158 link 132 network analysis 131 propagation 131 queuing 132 routing and hop counts 132 serialization 131 deleting handset record 72 deregister IP telephones 62 destination codes for fallback 84 network example 93 PSTN fallback 84 remote gateway destination digits 85 schedule 85 destination digits destination code 85 network example 93 remote gateway 78 destination gateway 140 destination IP network example 92 remote gateway 78 DHCP configuring 49 configuring for IP telephones 49 Invalid Server Address 50 IP telephone prerequisites 37 IP telephones 46 network prerequisites 30 VLAN on IP telephones 47 148 VLAN site specific options 149 dialed digits VoIP trunk routing 82 dialing plan CDP 93 116 destination code and destination digits 85 destination digits 80 MI ITG prerequisite 116 outgoing calls 76 PSTN fallback 81 system prerequisites 34 UDP 116 using UDP 86 Differentiated Service see DiffServ DiffServ 161 DISA VoIP trunks 73 display keys configuration 45 Distributed Host Control Protocol see DHCP DNs adding VoIP line pools 78 auto assign 34 auto assign IP telephones 47 before you start 69 changing handset name 72 H 323 terminals list 71 Hunt group target lines 89 NetVision 72 NetVision model 71 NetVision records 69 node range 92 records prerequisites 34 setting up target lines 89 documentation supporting 68 download firmware 60 staggered 61 DS30 split a
217. tion on that page must be the same on all Business Communications Managers in the network The Business Communications Manager supports only basic call to from NetMeeting S W version FP1 GA 2 5 GA 2 5 FP1 FAX over IP is not supported 3 0 2 5 FP Long tones do not work over IP trunks 2 5 FP1 MR1 1 3 0 2 5 FP1 Firewall Default Rules when enabled block call processing and signaling You must add an 2 5 FP1 MR1 1 additional rule to pass Protocol TCP UDP Destination Port H 323 for speech path to 3 0 initialize P0993474 03 Interoperability 151 Table 36 Software interoperability restrictions and limitations Software release Description of restriction limitation 2 5 FP1 2 5 FP1 MR1 1 3 0 If an IP Telephony Remote Gateway IP address is pointed at a Wan Link Interface which is a Published IP address the ISDN WAN Backup Feature will not support VoIP Traffic from any set type to that Published IP Address in some Network Topologies 2 5 FP1 2 5 FP1 MR1 1 3 0 Symbol portable IP handsets Login by Extension is login option offered by the telephone but is not currently supported by Business Communications manager The work around is to administer the extension as the username in Unified Manager The NetVision handsets do not support G 723 so they will be unable to negotiate a call on a VoIP trunk if the trunk is set to G 723 only Call Center ACD FEATURE 909 is not supported This is
218. to ensure that the system is correctly set up Some questions do not apply to all installations This guide contains a number of appendices that explain various aspects of the system directly related to IP telephony functions However refer to the Business Communications Manager Programming Operations Guide for specific information about configuring the data portion of the Business Communications Manager This section includes the following checklists e Network diagram on page 29 e Network devices on page 30 e Network assessment on page 30 e Resource assessment on page 31 e Keycodes on page 33 e Business Communications Manager system configuration on page 34 e IP telephones on page 37 Network diagram To aid in installation a Network Diagram is needed to provide a basic understanding of how the network is configured Before you install IP functionality you must have a network diagram that captures all of the information described in the following table If you are configuring IP telephones but not voice over IP VoIP trunks you do not need to answer the last two questions Table 1 Network diagram prerequisites Prerequisites Yes 1 a Has a network diagram been developed 1 b Does the network diagram contain any routers switches or bridges with corresponding IP addresses and bandwidth values for WAN or LAN links Also refer to Appendix D Interoperability on page 143 1 c Does
219. tworking 119 N name changing on handset 72 H 323 terminals list 71 H 323 Terminals record 70 NetVision 69 remote gateway 78 NAT network prerequisites 30 Netmask IP telephones 46 network prerequisites 29 NetMeeting choosing media type 145 configuring clients 98 supports slowStart 146 NetVision before you start 69 changing name for handset 72 common codec 150 configuration process 69 connectivity 67 deleting handset 72 DN records 72 H 323 Terminals record 70 installing 67 71 interoperability 143 model 71 name restrictions 69 serial cable 69 supporting documentation 68 unique name 70 updating H 323 record 71 network adjust jitter buffer 133 adjusting Ping measurements 157 analysing QoS needs 159 assessment prerequisites 30 asymmetrical media channel negotiation 145 devices prerequisites 30 DiffServ 161 implementing 131 insufficient link capacity 130 late packets sampling 157 link delay 132 P0993474 03 Index 177 loading 128 locations prerequisites 29 monitoring 162 planning modules 130 port settings 103 post installation measurements 134 quality of service 161 recording routes 156 reducing hop count 132 reducing packet errors 133 Sniffer 142 TCP traffic 160 traffic mix 160 troubleshooting routing 134 voice quality codec for IP telephones 42 networking additional feature configuration 127 Business Communications Manager prerequisites 34 call chain configuration 119 che
220. uding NetVision handset restrictions and Gatekeeper restrictions The tables are organized by Business Communications Manager software release numbers Table 36 Software interoperability restrictions and limitations Software release Description of restriction limitation All versions ITG payload sizes should be set to 30 ms All versions Silence suppression should be configured to the same value on both Business Communications Manager and ITG for example both on or both off Silence suppression is called Voice Activity Detection on ITG 2 03 GA M1 ITG interaction with more than one ITG when transferring conferencing working with 25GA two or more ITG cards they must be on the same subnet If they are not on the same subnet one way speech path situations can occur 2 5 FP1 The profile on the ITG must be set to the same first preferred codec as that of the Business 2 5 FP4 MR1 1 Communication software Software on the ITG trunk card must be 2 X 25 release 3 0 In order for features such as Transfer and Conference to operate correctly between all Business Communications Managers and ITGs in a network these are the rules e The First Preferred Codec for VoIP Trunks must be the same on all Business Communications Managers This is configured in Unified Manager under Services IP Telephony H 323 Trunks Media Parameters In addition if the first preferred codec is G 729 or G 723 the Silence Suppression op
221. unications Manager Telephony Features Handbook provides information about how to use Business Communications Manager call features The Business Communications Manager NetVision Phone Administrator Guide provides instructions for assigning features to the display list and includes an appendix containing a list of the features that work with NetVision handsets Configuring NetVision records This section provides the steps for configuring the various records that the NetVision telephone requires to work on a Business Communications Manager system This section describes e What information you require before you configure your handsets Gathering system information before you start How to set up an H 323 Terminals record on the Business Communications Manager to allow the NetVision handset to connect to the system Assigning H 323 Terminals records on page 69 Note DN records for NetVision handsets are created in the same way as for all other telephones on the system The various settings for DN records are described in the Business Communications Manager Programming Operations Guide Choose model IPWIs when configuring NetVision DN records P0993474 03 Chapter4 Installing NetVision telephones 69 Gathering system information before you start Ensure the following is complete or the information is on hand before you start configuring your NetVision telephones The Business Communications Manager has
222. uppression P0993474 03 Efficient Networking 125 Bandwidth requirements on full duplex links The following table provides bandwidth requirements for normal two way voice calls on a full duplex link for a variety of link protocols codec types and payload sizes Bandwidths for full duplex links are stated in terms of the individual transmit and receive channels For instance a 64 kbits full duplex link e g a DSO on T1 link has 64 kbits in the transmit direction and 64 kbits in the receive direction Table 30 Bandwidth Requirements per Gateway port for Full duplex links Ethernet B W PPP B W FR B W Payload No SP Silence No SP Silence No SP Silence Size Suppression Suppression Suppression Codec ms peak peak Avg peak peak Avg peak peak Avg Type kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s G 711 10 116 8 116 8 116 83 97 6 97 6 97 68 103 2 103 2 103 23 64 kb s 20 90 48 90 4 90 43 80 8 80 8 80 83 83 6 83 6 83 63 30 81 6 81 6 81 63 75 2 75 2 75 23 77 1 77 1 77 18 G 729 10 60 8 60 8 30 4 41 6 41 6 20 8 47 2 47 2 23 6 8 kb s 20 34 2 34 4 17 2 24 8 24 8 12 4 27 6 27 6 13 8 30 25 6 25 6 12 8 19 2 19 2 9 6 21 1 21 1 10 6 G 723 1 30 24 0 24 0 12 0 17 6 17 6 8 8 19 5 19 5 9 8 6 3 kb s G 723 1 30 22 9 22 9 11 5 16 5 16 5 8 3 18 4 18 4 9 2 5 3 kb s Notes 1 Gray background indicates payload sizes us
223. vices Server Type C Obtain a server address automatically Use the following server address information PAddess 47 3 11 85 Port C Name Cancel Apply Help P0993474 03 Chapter 3 Installing IP telephones 65 4 Enter the Published IP address of the Business Communications Manager in the IP address field 5 Inthe Port drop down menu select BCM 6 Select the Server Type tab Figure 15 i2050 Switch type x Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound Devices Server Type i2050 Software Phone Properties Meridian 1 C Centrex C CSE1000 C SL 100 C CSE6500 Enable s ymposium Cancel Apply Help 7 Click on the BCM option 8 Enable the Select Sound Devices tab for the USB headset To further configure this device through Unified Manager see Modifying IP telephone status settings on page 53 IP Telephony Configuration Guide 66 Chapter3_ Installing IP telephones P0993474 03 67 Chapter 4 Installing NetVision telephones This section describes how to configure the Symbol NetVision handsets to the Business Communications Manager system The information in this section includes e NetVision connectivity on page 67 e Configuring NetVision records on page 68 e Testing the handset functions on page 71 e Updating the H 323 terminals record on p
224. y cause them and how to troubleshoot the issue Table 12 IP telephone troubleshooting Problem Suggested solution or cause Telephone does not If an IP telephone does not display the text Connecting to server connect to system within two minutes after power up the telephone was unable to establish communications with the Business Communications Manager Double check the IP configuration of the telephone and the IP connectivity to the Business Communications Manager cables hubs etc Slow connection between f the connection between the IP client and the Business Communications the handset and the Manager is slow ISDN dialup modem change the preferred CODEC for Business Communications the telephone from G 711 to G 729 See IP telephone server Manager configurations on page 46 P0993474 03 Chapter 3 Installing IP telephones 49 Table 12 IP telephone troubleshooting Problem Suggested solution or cause One way or no speech paths Signaling between the IP telephones and the Business Communications Manager uses Business Communications Manager port 7000 However voice packets are exchanged using the default RTP ports 28000 through 28255 at the Business Communications Manager and ports 51000 through 51200 at the IP telephones If these ports are blocked by the firewall or NAT you will experience one way or no way speech paths Firewall note If you have the firewall filter set to P

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