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Avaya BCM 2.5 IP Telephony Configuration Guide

1. Net Link Mgr Alarm Service amp NAT D VPN Policy Management NTP Client Settings D a Management amp Diagnostics Ge Er 5 From the menu select Modify Codec Jitter Buffer The Terminal dialog box appears Figure 8 Terminal status dialog Property Sheet IP Terminal Status DN Read Only Field Status Online Type onn IF Address 17658026 Codec Defaut Fw Version 3002820 JitterBuffer Default T Terminal ID a Save Cancel FEAR ON Warming Applet Window 6 From the Codec menu select a Codec Specifying a non default CODEC for a telephone allows you to override the general setting You might for example want to specify a low bandwidth CODEC g 729 for a telephone that 1s on a remote or busy sub net 7 From the Jitter Buffer menu select a jitter buffer value For a telephone that has poor network connectivity to the Business Communications Manager increase the jitter buffer size P0937663 02 0 Chapter 3 Installing IP Telephones 43 Download Firmware to an i200X telephone Firmware 1s 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 to force immediate download to a telephone or if you
2. P0937663 02 0 Chapter 2 Prerequisites checklis t29 Table 4 Resource Assessment aaa 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 and has the DS 30 split been changed to 5 3 if necessary 4 c Have all necessary media resources for IP trunks clients vmail or WAN dialup been assigned or dedicated Keycodes All elements of VoIP trunks and IP telephony are locked by the Business Communications Manager keycode system Table 5 Keycodes LOC EE NN CNN 5 a Complete this question only if you are using VoIP trunks Do you have enough VoIP keycodes 5 b Complete this question only if you are using IP telephones Do you have enough IP client keycode Business Communications Manager System Configuration Several sections of the the Business Communications Manager must be properly configured prior to activation of IP telephony IP Telephony Configuration Guide 30 Chapter2 Prerequisites checklist Table 6 Business Communications Manager system configuration LO MEE NN 6 a Is the LAN functioning properly with the Business Communications Manager pf 6 b Is the WAN functioning properly with the Business Communications Manager NENNEN 6 c Has a dialing plan been created taking into account special considerations for IP telephony
3. To move an IP telephone and change the DN 1 Deregister the DN using the instructions in Deregistering DNs for online IP telephones on page 44 Disconnect the network connection and the power connection from the telephone Reinstall the phone at the new location For information on this see Connecting the 12004 Internet telephone on page 34 Configuring the i2050 Software Phone The 12050 Software Phone allows a computer equipped with a soundcard microphone and headset to function as an IP terminal on the Business Communications Manager system The 12050 Software Phone uses the computer s IP network connection to connect to the Business Communications Manager When you install the 12050 Software Phone on screen documentation walks you through the steps for installing the software To configure the 12050 Software Phone to connect to the Business Communications Manager 1 Follow the installation steps for installing your 12050 Software Phone until the software configuration utility appears or select the software configuration utility from your Windows Start Menu The utility opens to the Communications Server tab P0937663 02 0 Chapter 3 Installing IP Telephones 47 Figure 12 i2050 Communications server 12050 Software Phone Properties Ed Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound Devices 105 Switch Tupe C Use the following server address information
4. iz P Address Port 5000 C Mame 2 Enter the Published IP address of the Business Communications Manager in the IP address field Iin the Port drop down menu select BCM 3 Select the Switch Type tab Figure 13 i2050 Switch type 12050 Software Phone Properties Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound Devices 105 Switch Type 4 Click on the BCM option 5 If using a USB headset enable it in the Select Sound Devices tab To further configure this device through Unified Manager see Modifying settings for Nortel IP telephones on page 40 H 323 devices An H 323 device is any device that uses the H 323 protocol This is a protocol for Internet communications that uses audio video or data signals IP Telephony Configuration Guide 48 Chapter3 Installing IP Telephones Adding H 323 devices to the system The Business Communications Manager can support devices that use the H 323 protocol To install one of these devices 1 Inthe Unified Manager open Services IP Telephony and click on H 323 Terminals The H 323 terminal list appears Figure 14 H 323 Terminal list Configuration AASA H 323 Terminal List H 323 Terminal List oo ba BCh i 4r 65 862 131 D System Resources o Services Telephony Services S IP Telephony aE System Configurati Nortel IP Terminala H 323 Terminals S H 323 Tr
5. LAN CTE Configuration amp Console Service Voice Mail amp Voice Button amp DHCP amp DNS IP Routing amp SNMP 205 Monitor Web Cache Net Link Mgr Alarm Service amp NAT D VPN Policy Management NTP Client Settings D a Management amp Diagnostics 2 From the Published Address menu select the appropriate network interface Determining the published IP address Use the flowchart below to determine which card should be set as the published IP address The flowchart 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 on NAT see the Business Communications Manager 2 5 Programming Operations Guide P0937663 02 0 Chapter 4 VolP Trunk Configuration 53 Figure 17 Setting the Published IP address as the Published IP address Is NAT bled Set the network interface with gt o N the most anticipated VolP traffic Y Set the network interface on the private side as the published IP address Is the Business Communications Manager expected to connect to devices on the public side Are all of your public side Set the network interface on the devices using a VPN public sideas the published IP address Do you anticipate the most VolP Public traffic on you
6. To set the silence compression 1 In Unified Manager open Services IP Telephony and click on H 323 trunks 2 Click on the Media Parameters tab The Media Parameters dialog appears Figure 19 Media Parameters View Help 47 55 82 131 Comprehensive Local Gateway IP Interface Media Parameters Media Parameters os ial id se WW BCM 4765821315 System zu Tat Preferred Codec h 29 Y Resources o Services a 2nd Preferred Codec G 723 Y Telephony Services IP Telephony 24 3rd Preferred Codec G 71l ublaw 9 System Configuratil G711 aL x 9 Nortel IP Terminals o 4th Preferred Codec aLaw a H 323 Terminals aa Silence Compression Enabled O H323 Trunks co Remote Gatewi Jitter Buffer Voice AUTO PortRanges a amp Call Detail Recording LAN CTE Configuration o Console Service E Voice Mail Voice Button amp DHCP DNS S IP Routing D LANI D WANT D WAN 90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco web Cache 9 Met Link Mgr 3 Click the Silence Compression drop down menu and select either Enabled or Disabled If you select Enabled silence compression 1s only used when a G 729 or G 723 1 codec 1s in use Setting Jitter Buffers This section explains how to select the jitter buffer size used on VoIP trunks Jitter buffers are explain
7. Evening E Lunch P VoIP Sched 5 Sched 6 Scheduled Services amp System speed dial amp General settings Hunt groups amp Companion Hospitality Telco features Gr IP Telephony 5 Change Use Route to the route you configured for your PSTN fallback line 6 Under the Schedules list highlight VoIP P0937663 02 0 Chapter 4 VolP Trunk Configuration 67 Figure 33 VoIP schedule 47 65 82 131 z Comprehensive 3 8 YolP 6 VolP O QU BCM 47 65 82 131 amp System 2 Use route Route 02f Resources E Serices i Absorbed length an I 9 Telephony Services E System DNS amp Lines Restriction filters Call Routing Routes 24 z 9 Destination codes Schedules Normal Might fi Evening Lunch VoIP Sched 5 Sched 6 Scheduled Services amp System speed dial amp General settings amp Hunt groups Companion amp Hospitality Telco features Gr IP Telephony 7 Change Use Route to the route you configured for your VoIP line Configuring digits 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 inserting or remove absorbing digits For information of inserting and absorbing digits see the Business Communications Manager 2 5 Programming Operations Guid
8. Return to Menu Part No P0937663 02 Business Communications Manager 2 5 IP Telephony Configuration Guide NORTEL E NETWORKS Copyright 2001 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 All other trademarks and registered trademarks are the property of their respective owners P0937663 02 0 Contents PEGG AA AA AP D RENE dC ED ea eer 13 ere PIO oa eh AA aaa 13 Symbols used in this guide cc ee eee eens 14 Ee 4 5 a ER REX a a she eee hee ee ee 14 PO AD ohh Sods AA AA eee eae ees 15 Pee ee RP TTT T 1 7 HOW IO UAE TIBI ee ee edd 3d ee ee ee eee eee ee ee ee 18 Chapter 1 DID ie 9 CR or ee ee ee ee oe EI 19 IP telephones and VoIP trunks 0 ccc eee eens 19 lak 8D ecceuudea 3us6G4 68 AA gro nita seus b ends AT 20 Eo lE LC oc AA whee m 20 The ETO NEIWOIK 45 acce decade eddie E e e d e e e do oe e ed 2
9. 6 d Has a preference been established between pre configured DNs and auto assigned DNs If the preference is for auto assigned DNs complete 6 e 6 e Complete this question only if you are not using Auto assign DNs Have set DNs been programmed for the corresponding IP clients IP Telephones Complete this section only if you have IP telephones Table7 IP telephones Pew eM 7 a Are IP connections and IP addresses available for all IP telephones NENNEN 7 b If DHCP is not being used has all telephone configuration been documented and made available for telephone installers 7 c Has telephone powering been provisioned NENNEN 7 d Do computers that will be using the i2050 soft phone meet the minimum system requirements including headset 7 e Are IP connections and IP addresses available for all IP telephones NENNEN P0937663 02 0 31 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 If the destination is an IP telephone the arriving voice packets are converted to a voice stream which is routed to a speaker or headset If the destination is the Business Communications Manager the voice stream is routed to a circuit switched connection telephone or line or some form of gateway Note IP telephones require an IP network to reach the Business Communications Manager However t
10. Call Detail Recording Advertisement Logo Nortel Networks Default Codec AUTO T Default Jitter Buffer AUTO T D LAN CTE Configuration amp Console Service Cr Voice Mail Voice Button amp DHCP amp DNS Er IP Routing amp SNMP QoS Monitor Cr Web Cache Net Link Mgr D Alarm Service amp NAT D VPN Policy Management amp NTP Client Settings D a Management Diagnostics 2 Set Registration to ON to allow new IP clients to register with the Business Communications Manager Note For security reasons set the Registration to Off when you are not registering telephones 3 In the Password box type a password This is the password that installers must enter from the IP terminals to connect to the Business Communications Manager If this field 1s left blank no password 1s needed to register an IP terminal 4 Note The default password is bemi It can be changed to any alphanumeric string that is 10 characters in length or less 4 Set the Auto Assign DN box If Auto Assign DNs is set to ON the Business Communications Manager system assigns a free DN to a set being registered instead of prompting the installer for the set DN If registration and Auto Assign DNs are both set to ON and the Registration password is blank first time connected IP clients will be assigned a DN without requiring installer intervention The system selects this
11. Deregister DN System Configuration Nortel IF Terminals H 323 Terminals H 323 Trunks PorRanges Call Detail Recording D LAN CTE Configuration Console Service amp Voice Mail Voice Button DHCP DNS IP Routing amp SNMP amp QoS Monitor amp Web Cache Net Link Mgr D Alarm Service NAT amp VPN Policy Management NTP Client Settings D a Management amp Diagnostics 4 Press Enter In the Deregister DN field type the number of the DN you want to deregister The DN is deregistered IP Telephony Configuration Guide 46 Chapter3 Installing IP Telephones Moving IP telephones IP telephones retain their DN when they are moved to a new location The following instructions apply to I200X phones computers with the 2050 softphone installed and H 323 devices To move an IP telephone without changing the DN Disconnect the power from the IP telephone or 3 port switch Disconnect the network connection At the new location reconnect the network location and the power connection Aa sc N If the new location is on a different LAN or WAN from the old location the subnet mask default gateway IP S1 IP and S2 IP may change If this is the case you must change the settings for the telephone To do this see Connecting the 12004 Internet telephone on page 34 Do not change the Set IP Address
12. e Installing keycodes e Configuring media parameters e Outgoing call configuration e Incoming call configuration Note If you are using the Business Communications Manager with an M1 you must set up the system to be compatible with the M1 See Appendix D Interoperability on page 111 Installing keycodes Before you can use VoIP you must obtain and install the necessary keycodes See the Keycode Installation Guide for more information Setting the 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 anda WAN interface WANI that 1s connected to the public network thenW ANI should be set to the published IP address To set the published IP address IP Telephony Configuration Guide 52 Chapter4 VolP Trunk Configuration 1 In Unified Manager open Services and click on IP Telephony The Global settings tab appears Figure 16 Global IP settings 47 55 82 131 z BANG Ne Global IP Setting Global IP Setting Gl ACM 4765221315 5 amp System Published IP Address IP LANT Resources al Q Services pss D Telephony Services o IP Telephony System Configuratian e H 323 Trunks Gr PorRanges Call Detail Recording
13. 0 Figures Figure 1 Figure 2 Figure 3 Figure 4 Figure 5 Figure 6 Figure 7 Figure 8 Figure 9 Figure 10 Figure 11 Figure 12 Figure 13 Figure 14 Figure 15 Figure 16 Figure 17 Figure 18 Figure 19 Figure 20 Figure 21 Figure 22 Figure 23 Figure 24 Figure 25 Figure 26 Figure 27 Figure 28 Figure 29 Figure 30 Figure 31 Figure 32 Figure 33 Figure 34 Figure 35 Figure 36 Figure 37 Figure 38 Figure 39 Figure 40 pie APA APA AA 21 Set registration properties aa 32 AC NA AA ee 38 Bi AA PAA 39 BA PA AA AA 40 IP Terminal SUB aue ru RRmLha KANAN shes seeesenee ands enue 41 Configuration menu 1 tenes 42 Pe SIE o Errem 42 Naik a lan ODE AA 43 IP Terminal Status ee ee ee ee GA kiami 44 Deregister Offline DN 0 2c eas 45 i2050 Communications Server cee eee 47 Paba UAAP 47 Ho TUWA amak deeds eens LANG BESH bocca bens bees 48 H 323 Terminal list dialog llle 49 Clee BE SOLIDE us sais besu AA AA AA 52 Setting the Published IP address 00 0 53 Media parameters anaana aaa 54 bist DID asd BR GA KONG ee tanse eae PGKA 55 Media parameters sellers 56 TUNE LNE A PAA AA ERE d 57 Eoaea P APAPAP 58 Remote gateway list 0 ee ee ee eee eee 59 Remote gateway dialog 0 ccc ee eee eens 60 PSTN Geille ae cele oe AA 61 VOIP SONGU uoi oooh RSG SSAA ESS ASSES EES EGS ERATE ROSES 62 VOIP HOLUUnO SOVICO 64465454 seb
14. 0 0 00 cc ee eee 96 Setting Non linear processing 0 ee eae 96 Determining network loading caused by IP telephony traffic 96 Other intranet resource considerations 0 000 cee eee eee 99 implementing the DON DER ccs wk cade UE dee eee OR ICE CORE CE OR EE ORE 99 Fogo LIE TIT C 2055 77157 077 1217 0012257 11211702773 99 Getting the best results from your system llle 99 Further network analysis uasa ee ceed eee Weed deed EE d RR EK WAG NGA 99 Components of delay uu BA eed cw ced nosh dno mee 9 RR SEG ERG Goa d 99 aute p Etc A ooo bere de bE eee eee eee Rees 100 Hnegucmna NOP COUN acus grace dw kA OGOUE A been bbe PA RR e dox era 100 ROUINO wawa PTT PP 102 Post installation network measurements eens 102 Appendix B Silence COMDICSSION isis uia och ae PG AKA BEKE KR EB AB OM 103 Silence compression on Half Duplex Links a 103 Silence compression on Full Duplex Links 0000 e eee eee 105 OUI s e Fis M P 108 PIXGPEREESS Angra SAU ERPEESIRASNEGUEREE S PARVA SEU E E ENE 108 Appendix C Network Performance Utilities lerne 109 gitar ere eS eee yee ee KO Tee ee eee ee ee JURE EROR eee ee UR HO ee E 109 IIBOBIOUIB 4 o2aceccecacteesoauswauese eee nw ates SESASSU dd Rad aia qa 109 c p AA NITET PTT 109 Appendix D lucis piel C UU 111 Speech Path Setup Methods 4 isa ac dc ora pd XC OR 9 Yn CIR e E Ra
15. 9c 303 OR HEC MA PAA oe ae RN 77 LOO I U et ETET SORGE 9B ee T 78 Port settings for legacy networks llle 79 Using a gatekeeper llle han 80 The call signalling method 6 4 4 4440 cess a3 hk a RA ee RR e ORARE REG 80 PUPS Tees 2Z2 x224295235 52292932 2 9 2 2 2 2 duq UE QM RAM LAGA a0 Modifying the call signalling method llle 81 Gatekeeper call scenarios llli 82 Chapter 5 Typical BODNCAUGIIG p usa deae acm ede ew de BG eee eee ee Rm cR KA 85 M1 to Business Communications Manager over IP trunks 85 PILOT COON uiis aad bade dida aA PRAES REGE nena ee eee ERN do 86 Multiple Business Communications Manager systems et DE ho ks TTE 86 Multi location chain with call center llle 87 Business Communications Manager to IP Telephones 89 Appendix A HOPED NBI AAP AP RARO CER edo diee oe ac ed dd dr 91 Determining the bandwidth requirements llle 91 Determining WAN link resources llle 91 RW OU m 91 Network Engineering dd dd BKA 049 Ed ER EE EE E EHE E ERE EH 92 Bandwidth Requirements on Half Duplex Links llle 93 Bandwidth Requirements on Full Duplex Links ls 94 LAN Engineering Examples eleele es 95 AE 64 06464 mtd 3 3 CE do qd aci Ea doe AY 96 IP Telephony Configuration Guide Additional feature configuration 0
16. Business Communications Manager media parameters table Table 17 on page 113 lists the names used by the Business Communications Manager local gateway table and the matching names in NetMeeting Table 17 Name comparison Business Communications Manager media parameters table MS NetMeeting G 723 1 6 3 Kbit s MS G 723 6400 bit s G 711 u law CCITT u law G 711 A law CCITT A law No feedback busy station G 723 1 5 3 Kbit s MS G 723 5333 bit s The Business Communications Manager VoIP gateway provides call progress tones in band to the user If a 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 IP Telephony Configuration Guide 114 Interoperability Symbol Telephones The Business Communications Manager currently can only receive slowStart calls from a Symbol phone Since the Symbol phones default to use H 323v2 fastStart they must be changed to use slowStart There are two methods that may be used to do this 1 Manually edit the files created by NetVision Phone Administrator NVPA program Add a configuration parameter to set the phones to use slowStart methods 2 Log into the phone as a privileged user and set the call setup mode to slow start See the Symbol documentation for mo
17. CODECS e The G 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 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 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 1s However voice latency delay also increases Latency is very problematic for telephone calls as it increases the time between when one user speaks and the user at the other end hears the voice The administrator can adjust the default jitter buffer size to the following values e NONE Minimal latency best for short haul networks with good bandwidth e AUTO Business Communications Manager will dynamically 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 IP Telephony Configurat
18. Detail Recording D LAN CTE Configuration Console Service Voice Mail Voice Button amp DHCP amp DNS IP Routing amp SNMP 205 Monitor amp Web Cache Net Link Mgr Alarm Service amp NAT D VPN Policy Management NTP Client Settings Management amp Diagnostics Online 2004 3 Select the IP Terminal that you want to change the properties for Summary General IP Terminal Status Deregister Offline DN 4765 80 26 Default 4 Open the Configuration menu or right click anywhere on the listing for the terminal to bring up the menu IP Telephony Configuration Guide 42 Chapter3 Installing IP Telephones Figure 7 Configuration menu Configuration Performance Fau Heport loo Coco ew Help ban aehenana IP Terminal Status Deregister Offline DN Add D Dal A IP Terminal Status 9 tal BCM 47 55 82 131 DN NESTE ees LEESON amp System Online anna AP CRON Maka 3002620 Default Resources Deregister DN 7 a Services Force firmware download amp Telephony Services Modify Codec JitterBuffer S IP Telephony System Configuration Mortel IP Terminals H 323 Terminals H 323 Trunks Cr PorRanges Call Detail Recording D LAN CTE Configuration amp Console Service Voice Mail Voice Button amp DHCP amp DNS IP Routing amp SNMP 205 Monitor amp Web Cache
19. Duplex Link Without Silence compression TX Rx Tx Rx Chan Bandwidth Conversation Hello Fred This is Susan Do you have a minute Fred here Hi Sure Bandwidth used Channel Link max Voice frames sent even when speaker is silent When silence compression is enabled voice packets are only sent when a speaker 1s talking In a typical voice conversation while one speaker 1s talking the other speaker is listening a half duplex conversation Figure 49 on page 104 shows the peak bandwidth requirements for one call a half duplex link with silence compression enabled Because the sender and 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 49 One Call on a Half Duplex Link With Silence compression TX Rx Tx Rx Chan Bandwidth Conversation JHello Fred This is Susan Do you have a minute Fred here Hi Sure d Bandwidth used Channel Link max V ee re Time Half dulpex call alternates use of half duplex bandwidth P0937663 02 0 Silence compression 105 The affect of silence compression on half duplex links is therefore to reduce the peak and average bandwidth requirements by approximately 50 of the full transmission rate Because the sender and receiver are sharing the same bandwidth this affect can b
20. Most of corporate intranet traffic is TCP based Different from UDP that 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 ofWAN links e IP telephony traffic streams are unfairly affected 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 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 When received the WAN link marks the H 323 flows as Premium traffic and places the flows in the hi
21. OS 112 POP 2 s drd 3 873 ROC ud dea n o cda Ce AC e Oeo Sess ROS abe V race as 112 kusa uasa Peces Rese 444 ees P15z4bg ARUM ICE NE QURE MORS Void qos 113 Asymmetrical Media Channel Negotiation 0 2 0 0 cece ees 113 No feedback busy station cette 113 GUI Telephones 5 gga hee do oid 5 694 3 40 994 ore AA 114 Appendix E Quality of Service ad oe ace RR RR AA PAA 115 condi cph a rrr 115 WEES UNO Meet LO uaaerdadsriste ras REA REG ERESXSdadX YE EE 116 Measuring end to end network delay aasa aaa 116 Measuring end to end packet loss 0 0 cee ee ee 117 P0937663 02 0 pce OOS a ee ee 1212435 117 Adjusting ping measurements leere 118 Measurement procedure a 118 Other measurement considerations 2 0 0 0 ee eee 119 Implementing QoS in IP networks wii cee wee dog kom RC KR a ea ds 120 Tie A Loa dh hak eos ee ee ee a ee 8 ANNA 120 TOP Wane DONG 2c cuca ska KA ceed sr iid ceauskecs o4ee486i4 28 121 Business Communications Manager Router QoS Support 121 Network Quality of Service 0 eee 121 Network Monitoring eed 4 ERR RR dae dd ed d e n dde dee eR Re PE Rd 122 Quality of Service parameters llle 122 Fallback to PSTN 2 6c RR nmn 123 SOO KA c ean ed E phe ONO Oe GAUGE KG E ee PAA NG 125 lie ERPAT be 4 on eee ae hw SEES hee oe ode Oe eae 7 5 127 IP Telephony Configuration Guide P0937663 02
22. callers dial 55 to call phones on the Ottawa Business Communications Manager after dialing the destination digit The installer sets up a schedule called VoIP with these settings e Service Manual e Overflow Y The installer defines a new route called Route 003 and sets it to use PRI A 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 7 e Under the Normal schedule the installer assigns Route 003 which uses PRI A e Under the VoIP schedule the installer assigns Route 100 which uses the VoIP lines in line pool O The installer configures the call digits They must be configured so that calls will still be processed correctly if routed over PSTN For information about configuring dialing plans see the Business Communications Manager 2 5 Programming Operations Guide 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 The installer programs target line 241 with the received digits 321 then assigns this target line to DN 321 This means that calls received containing the digits 321 are delivered to DN 321 The installer repeats this procedure for each telephone using different received digits and target lines Making calls From a set on Business Communic
23. 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 Table 19 on page 117 you can calculate the load of voice traffic per link Table 19 Site pairs and routes Site pair Intranet 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 IP Telephony Configuration Guide 118 Quality of Service Table 20 Computed load of voice traffic per link Links Traffic from Santa Clara Richardson Santa Clara Richardson Santa Clara Tokyo Santa Clara Richardson Richardson Ottawa Santa Clara Ottawa Santa Clara Tokyo Santa Clara Tokyo Richardson Ottawa 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 needs additional adjustments by adding 140 ms to explain the processing and jitter buffer delay
24. compression 1s active The source gateway sends information packets to the destination gateway informing it that silence compression 1s active and describing the background comfort noise to insert The source gateway only sends the information packets when it detects a significant change in background noise P0937663 02 0 109 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 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 O 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 datag
25. eee Hee ea KABA EARS ER TERE ERO 63 Fee anak axed nena TEKA hes oem ayeaees owes DB UKG eee ae ase 64 poe ake Licda rmm 64 BOE RUSO as 3 392 04199 015595919 AA 65 Add destination code dialog liliis 65 NGAGE Ah us aaa d lcd de de die dede dae dede eq e ORES 66 VOF SABES kb eo 4 8c 5 oh eh OS 5454 AT 67 Remote Gateway list 0 00 0 cee eas 69 Remote Gateway dialog 00 ccc eee eens 69 Rennie PONV fallback AA 71 NetMeeting options pA KKK KKK brad VEG dd hoe 4 eee d ERROR 76 NetMeeting advanced options 0 0 cee eee 76 ee acras dokaeg ee a IE dedic 303 wd A he Oe ee Oo di 78 POM INIDES DIR amak BNG 65466 bonds 6454544566 dgra aE cause 79 IP Telephony Configuration Guide 10 Figure 41 Figure 42 Figure 43 Figure 44 Figure 45 Figure 46 Figure 47 Figure 48 Figure 49 Figure 50 Figure 51 Figure 52 Figure 53 Figure 54 P0937663 02 0 Local gateway IP interface AG eo S EOD RES DOOD ORC 81 Business Communications Manager systems with a gatekeeper 82 M1 to Business Communications Manager network diagram 85 Multiple Business Communications Manager systems network diagram 87 M1 to Business Communications Manager network diagram 88 M1 to Business Communications Manager network diagram 89 Calculating network load with IP telephony traffic 97 One Call on a Half Duplex Link Without Silence compression 104 One Call on a Half Duplex Link With S
26. 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 1s received at the next hop router is the link delay Methods to reduce link delays are Upgrade link capacity to reduce the serialization delay of the packet This also reduces the utilization of the link reducing the queueing delay Before 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 acrossWAN links Some of the ways to reduce hop count include P0937663 02 0 Networking 101 e Improve meshing Add links to help improve routing adding a link from router to router4 instead of having the call routed from router 1 to router 2 to router 3 to router 4 reducing the hop count by two e Router reduction Join co located gate
27. gateway for the Santa 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 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 44 This means that callers dial 44 to call phones on the Santa Clara Business Communications Manager after dialing the destination digit The installer sets up a schedule called VoIP with these settings P0937663 02 0 10 11 12 13 Chapter 4 VolP Trunk Configuration 73 e Service Manual e Overflow Y The installer defines a new route called Route 003 and sets it to use line pool PRI A 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 7 e Under the Normal schedule the installer assigns Route 003 which uses line pool PRI A e Under the VoIP schedule the installer assigns Route 100 which uses the VoIP lines in line pool O The installer configures the call digits They must be configured
28. intranet e asymmetrical routing Post installation network measurements The network design process 1s 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 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 conform to internal quality of service standards and that QoS objectives are always met P0937663 02 0 103 Appendix B Silence compression Silence compression reduces bandwidth requirements by as much as 50 This appendix explains how silence compression functions For information on enabling silence compression n VoIP gateways see Setting Silence Compression on page 54 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 technolo
29. location to a data processing device at another location enbloc All dialed digits sent in a single expression The system waits for all digits to be dialled before processing the call 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 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 IP Telephony Configuration Guide 126 Glossary H 323 The ITU standard for multimedia communications over an IP network Enterprise Edge IP Telephony supports H 323 Hub Center of a star topology network or cabling system kbit s kilobits per second Thousands of bits per second Latency The amount of time it takes for a discrete event to occur Mbit s Megabits per second Millions of bits per second Modem 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 Random electrical signals generated by circuit components or by n
30. m System SOnng ital zE 4th Preferred Codec G 7ll aLlaw a Mortel IP Terminala i 9 H 323 Terminals 24 Silence Compression Enabled og H323 Trunks 225 Remote Gatew Jitter Buffer Voice AUTO amp PortRanges 24 Call Detail Recording 89 LAN CTE Configuration amp Console Service a Voice Mail amp Voice Button amp DHCP DNS e IP Routing Cr LANI D WANT D WAN V90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco Web Cache Net Link Mgr 3 Click the First Preferred Codec menu Select the codec you want to use as the first preferred codec This is the most preferred codec to be used on VoIP trunks 4 For each preferred codec use the process described in step 3 Setting Silence Compression This section explains how to set silence compression on VoIP trunks The silence compression feature identifies periods of silence in a conversation and stops sending IP speech packets during those periods In a typical phone 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 s end This greatly reduces bandwidth use P0937663 02 0 Chapter 4 VolP Trunk Configuration 55 G 723 1 and G 729 support silence compression If a conversation is using G 711 silence compression does not occur
31. number from the Norstar Digital Telephone Range 5 Inthe Advertisement Logo box type a string of text characters This message is displayed on the first line of the IP terminal s display This string can be a maximum of 24 characters P0937663 02 0 Chapter 3 Installing IP Telephones 33 6 From the Codec menu select a default Codec or leave the Default Codec at Auto This is the Codec that is used if an IP telephone has not been configured with a preferred codec For information on this see Choosing a codec on page 33 7 From the Jitter Buffer menu select a Jitter Buffer level For information on choosing a Jitter Buffer see Choosing a Jitter Buffer on page 33 Choosing a codec The default codec is used when an IP client has not been configured to use a preferred Codec see 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 goes on a call For example if both endpoints of the call are I20X X 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 For IP telephones the Business Communications Manager supports both A and MU law variants of the G 711 CODEC as well as the G 729 and G 723
32. 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 1s overloaded the router 1s continuously processing intensive tasks Processing intensive tasks prevents the router from forwarding packets To correct this reconfigure or upgrade the router Routers 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 IP Telephony Configuration Guide 102 Networking Saturation refers toa situation where too many packets are on the intranet Packets can be dropped on improperly 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 101 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 e routing instability e wrong load splitting frequent changes to the
33. service e no network 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 1s active only at call setup A call in progress does not fall back 1f the quality degrades Calls fallback if there 1s 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 124 Quality of Service P0937663 02 0 125 Glossary backbone A network s major transmission path 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 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
34. sure the M1 ITG meets the following requirements e ITG Kit NTZC44BA Delta 24 24 e RIs25 30 e S W Packages 57 58 59 145 147 148 160 2 Setup the M1 ESN programming CDP UDP For information on this see your M1 documentation 3 On the Business Communications Manager 2 5 Unified Manager e Set up outgoing call configuration for the VoIP gateway e Set up a remote gateway for the Meridian 1 IP Telephony Configuration Guide 86 Chapter5 Typical applications e Set phones to receive incoming calls through target lines e Configure the PSTN fallback and enable QoS on both systems e Activate and program CDP or UDP For information on CDP and UDP see the Business Communications Manager 2 5 Programming Operations Guide MCDN functionality To use MCDN functionality over PRI set up e MCDN PRI settings on the MI For information on this see your M1 documentation e SL 1 MCDN keycodes on the Business Communications Manager 2 5 Multiple Business Communications Manager systems across VoIP trunks This system allows multiple offices 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 Users have full access to system users applications PSTN connections Unified Messaging The network diagram shows two Business Communications Managers but addi
35. suspect that a particular telephone has corrupted firmware To download firmware 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 Figure 9 IP Terminal Status Configuration View E e Summary General IP Terminal Status Deregister Offline DN IP Terminal Status 9 GJ BCM 47 55 82 131 DN amp System 388 Online i2004 47 65 80 26 Default 3002620 Default Resources z o Services D Telephony Services S IP Telephony System Configuratian Martel IP Terminals H 323 Terminals H 323 Trunks Gr PorRanges Call Detail Recording LAN CTE Configuration amp d Console Service Voice Mail Voice Button amp DHCP amp DNS IP Routing amp SNMP 205 Monitor amp Web Cache Cr Net Link Mgr Alarm Service amp NAT D VPN Policy Management NTP Client Settings D a Management amp Diagnostics 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 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 IP Telephony Configuration Guide 44 Chapter3 Install
36. the keyword NAME e Transport Address IP Address Identified by the keyword TA In the following example the Business Communications Manager is assigned an E 164 and an H323 Identifier alias Alias Names tel 76 name bcm10 nortel com Modifying the call signalling method To modify the call signalling method 1 Inthe Unified Manager open Services IP Telephony and click on H 323 trunks Figure 41 Local gateway IP interface View Help ABR ashton 2 Local Gateway IP Interface Media Parameters a idd 3 el A Local Gateway IP Interface T iw BCM 47 55 82 131 3 amp System 7 Fallback to CircuitSwitched Enabled All Resources 2 O Services M E Call Signaling Direct Telephony Services E TAGE Ss 9 IP Telephony 2 jE GatekeeperIP hons CO System Configuratii Nortel IP Terminalsc Mars aene AN H 323 Terminals d H323 Trunks 24 Remote Gatew RE EFM eee eee eee ees amp PorRanges oa Call Detail Recording LAN CTE Configuration amp Console Service E Voice Mail Voice Button amp DHCP DNS S IP Routing D LANI D WANT D WAN 90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco web Cache 9 Met Link Mgr 2 Inthe Call Signalling list select the appropriate setting For information about the settings see The call signalling method on page 80 If selecting GateKeeper
37. you want to set up 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 threshold 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 60 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 Table 9 shows a sample QoS Monitor Table 9 QoS status a ee G 729 kbit s kbit s QoS L monnor Note For the QoS monitor and PSTN fallback to function both Business gt 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 Port settings In some installations you may need to adjust the port settings for the Business Communications Manager can work with other devices IP Telephony Configuration Guide 78 Chapter4 VolP Trunk Configuration Using firewalls Firewalls can interfere with co
38. 0 Business Communications Manager eee eee eee eens 21 LILA eee eo PEE ee ee ee ee ea re eee OPAPP PA ene eer a ae ere A 22 ITOT c45445b54boee eek eh ss 5555560545505 56455555854 nka NNi 22 Traditional telephones 44464656 s dos 4454 6664045 FREE ER EHH EE EES 22 Labi pA AASA 22 PIG 25 BAGA 550 E 1x4 Eq dd 545 48 S44 GAGA KIS 22 UG ATCP 22 I TES ao BA ee a ees A HOT Oe eee NEN eee ees 22 id RA AA PN PA AA 23 E KABANG LEG ee KIA EEEE EES GB BAG KAG KARA ee ee ae EE ee ee ae 23 Public Switched Telephone Network aaa 23 Key IP Telephony Concepts 6 654466585805 KA 005544564444 E ER RES 23 deal ko oo ENEE ee be ee ok hee PAA 23 REO ES 445 044 AA 24 ES I o cen otn mp nea ph deb Oe a hen e d PE Sarda Roper pa de door ke 24 Chapter 2 Prerequisites checklist Leere 27 tct es cs DANPETITITITITITITITIOTIOTIT IT OT TT TIT IT TT TTTTTTTTTTTTTTT 27 bs 4 AA 27 Network assessment 0000 eee eee eee 28 Resource Assessment ohh 4 pa dn X CICER aE C eb ERE deed RE aK KA 28 PO oo ed boa bod po obo SS ESE PSE EKO NENA qe PO HAS 29 Business Communications Manager System Configuration 29 lagi te AA AA AA AA 30 IP Telephony Configuration Guide Chapter 3 Installing IP Telephones 0 0c eee es 31 Preparing your Business Communications Manager for IP telephone registration 31 Ee ee Greate nd dedo qaod nes eee di dod eee eh oe Jer pd EP 33 Choosing a Jitter Buf
39. 0937663 02 0 Chapter 4 VolP Trunk Configuration 79 Figure 40 Port ranges dialog i Property Sheet PortRanges a ee Begin o aaa ESSE Ready Warming Applet Window 4 Change the port settings 5 Click the Save button Port settings for legacy networks Business Communications Manager 2 5 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 applications 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 2 5 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 2 5 re
40. 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 IP Telephony Configuration Guide 74 Chapter4 VolP Trunk Configuration 10 11 12 13 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 321 to 331 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 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 55 This means that
41. 82 131 1s the published IP address Figure 3 Global options View Help ippo ohenseb Summam DHEP Mode Global Options r Global Options tal BCM 4T 55 82 1 amp System IR DomeinName miai etene amp Resources Services 2 HHHHHI Telephony WINS Node Type 0146 ERR IP Telephon Call Detail H MORTEL IP Terminal Information 128 ij oreki2004 2 47 85 82 131 7000 1 250 47 65 22 131 7000 1 250 LAN CTEC Console S8 Voice Mail NTP Client amp 3 Management Diagnostics 3 Click the Summary tab The summary screen appears P0937663 02 0 Chapter 3 Installing IP Telephones 39 Figure 4 DHCP summary View Help 47 65 82 131 Comprehensive GJ BCM 47 55 82 131 amp System Description ECM DHCP S Resources erver Services BAR Telephony Services EX D IP Telephony Call Detail Recording Status Disabled v LAN CTE Configuration Console Service Voice Mail Voice Button o DHCP Local Scope Remote Scope amp DNS Gr IP Routing amp SNMP 205 Monitor Web Cache Net Link Mgr amp Alarm Service amp NAT D VPN Policy Management NTP Client Settings D a Management amp Diagnostics 4 Set the Status box to Enabled 5 Open Services DHCP Local Scope LANX where LANX is a LAN that contains IP sets that
42. E i 2 Enter a unique access code for this line pool Ensure that no other line pools use this access code Configuring DNs to access the VoIP line pool Each DN that is to use the VoIP lines must be authorized to use the VoIP line pool P0937663 02 0 3 4 Chapter 4 VolP Trunk Configuration 59 In Unified Manager open Services Telephony Services System DNs Assigned DNs DN XXX Line Access and click Line Pool Access DN XXX is any DN that you want to allow to use VoIP trunking Click Add The Add Line Pool Access dialog appears Type the letter of the VoIP line pool Click Save Repeat this procedure for every DN 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 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 Figure 23 Remote gateway list Configuration View Help 47 65 82 131 E sate Remote Gateway Add jl Remote Gateway T ld BCh C47 65221315 ap amp System Resources o Services Gr Telephony Services e IP Telephony a System Configuratii Nortel
43. IP Terminala 5 H323 Terminals e H 323 Trunks 2 Remote Gatewi PortRanges E Call Detail Recording 53 LAN CTE Configuration amp Console Service a Voice Mail Voice Button amp DHCP 9 DNS e IP Routing D LANI D WANT D WAN 9 90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco Web Cache 9 Met Link Mgr Pia I DEESSET SSES I DP cd pcalbcm19 47 65 80 43 Enabled 0 0 0 0 Voice Met 7 IP Telephony Configuration Guide 60 Chapter4 VolP Trunk Configuration 2 Inthe Configuration menu select Add a new entry The Remote Gateway window appears Figure 24 Remote gateway dialog Property Sheet Remote Gateway Name abemt8 0 Read Only Field Destination IP 47 65 80 43 QoS Monitor Enabled Transmit Threshold CY Receive Threshald Gateway Type Destination Digits E Warming Applet Window 3 Inthe Name field type a name for the remote system 4 In the Destination IP field enter the IP address of the system 5 In the Destination Digits field 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 6 Set the QoS monitor option If you intend on using fallback to a PSTN line s
44. P 35 38 ethernet connection 34 firmware downloading 43 installing 31 Invalid server address 36 Jitter buffer 33 modifying 40 New Set 36 No ports left 36 ports 35 registering 31 registration 32 Registration disable d36 Set IP viewing 36 setIP 35 settings modifying 40 subnet mask 35 Troubleshooting 36 IP Telephony Benefits 19 concepts 23 Introduction 19 IP telephony networks 20 IP Telephony network DHCP 38 IP telephony network checklist 27 J jitter 123 IP Telephony Configuration Guide 128 Index Jitter buffer defined 24 IP sets 33 jitter buffer size 101 L link delay 100 M M1 ITG defined 22 Interoperatibility 111 Measuring Intranet QoS 116 Media path redirection 111 N network devices checklist 28 network loading 96 network measurements 102 P Packet delay 122 packeterrors 101 Packetlos s122 Ping 109 ping 118 ports IP set 35 prerequisites 27 Q QoS 120 defined 24 QoS Monitor 77 Quality of Service Monitor 77 R Registration IP sets 32 Routing and hop count 100 routing instability 102 P0937663 02 0 S speech path setup methods 111 subnet mask IP sets 35 Symbol interoperability 111 Symbols 14 T text conventions 14 tip 14 Troubleshooting IP sets 36 V VoIP trunks defined 19 W Warning symbol 14
45. P schedule e Configuring routes e Creating a destination code e Configuring digits e Activating the VoIP schedule e Turning on QoS monitor Setting up a VoIP schedule Setting up a VoIP schedule allows you to create two different call routes for a single destination code One of these is a VoIP route The other is typically a PSTN route The PSTN route uses the Normal schedule which runs when no other schedule is activated To set up the VoIP route 1 Inthe Unified Manager open Services Telephony Services Scheduled Services Common Settings Schedule Names IP Telephony Configuration Guide 62 Chapter4 VolP Trunk Configuration Figure 26 VoIP schedule 47 65 82 131 Comprehensive 2 Schedule 4 Schedule 4 eG Gil ACM 4765221315 dela lalalalala Resources E O Services e SE E Telephony Services S amp System DNs Lines Restriction filters amp Call Routing Scheduled Services amp Ringing service Restriction service 2 amp Routing service Common settings Schedule name 4 Schedule 0 F Schedule 1 8 Schedule Schedule 3 by Schedule 4 Schedule 5 Schedule 6 Schedule times System speed dial General settings Hunt groups Companion Hospitality Telco features Gr IP Telephony Select a schedule that will not be used for another purpose typically Schedule 4 Change the schedule name to VoIP Open Services Telephony Servic
46. Routed or GateKeeperResolved in the Gatekeeper IP box type the IP address of the machine that is running the gatekeeper IP Telephony Configuration Guide 82 Chapter4 VolP Trunk Configuration If selecting GateKeeperRouted or GateKeeperResolved in the Alias Names box type one or more alias names for the gateway For information on setting alias names see Alias names on page 80 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 Figure 42 on page 82 shows a network with three Business Communications Managers and a gatekeeper Figure 42 Business Communications Manager systems with a gatekeeper gatekeeper IP 10 10 10 17 BCM Ottawa IP 10 10 10 18 BCM Santa Clara IP 10 10 10 19 BCM 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 signalling 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 2 The gatekeeper resolves DN 421 to 10 10 10 19 and returns this IP in an AdmissionConfirm to the Business Communications Manager Ottawa 3 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 signalling 1s set to Gatekeeper Routed
47. Threshold LY Receive Threshold 2 HAHA Gateway Type Destination Digits fooon AHHH TELE EY Se Warming Applet Window For the QoS Monitor field select Enabled Set the Transmit Threshold and Receive Threshold to a value between 0 and 5 This marks the level of quality that the gateway 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 IP Telephony Configuration Guide 70 Chapter4 VolP Trunk Configuration communications in one direction are more important than in the other direction you can set up asymmetrical thresholds For information about using the QoS monitor see Quality of Service Monitor on page 77 Enabling PSTN fallback To enable PSTN fallback 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 TDM 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 Incoming call configuration This section explains how to receive incoming calls placed over a VoIP network For information about setting up your Business Communications Manager to place outgoing VoIP calls see Outgoing call
48. a voice conversation gaps in the conversation represent packet losses Some packet loss less than 5 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 P0937663 02 0 Quality of Service 123 Delay variation jitter The amount of variation in packet delay 1s 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 reroutes 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 e bad network conditions e the remote gateway is out of
49. age will remain on the display until a port becomes available and the telephone is powered down then powered up To obtain more ports you may need to install additional 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 e NEW SET The telephone has not been connected to the Business Communications Manager before and must be registered P0937663 02 0 Chapter 3 Installing IP Telephones 37 e Registration Disabled The Registration on the Business Communications Manager is set to OFF Note To see the configuration information of a telephone connected to the gt Business Communications Manager When the telephone is not on a call press the EXPAND key blue key at the bottom right of the telephone followed by the SERVICES key key with image of a globe To see the Codec data for a telephone while it is on a call Press the EXPAND key blue key at the bottom right of the telephone followed by the SERVICES key key with image of a globe Use the following strategies for troubleshooting an IP telephone e If an IP telephone does not boot use VNC to access the Business Communications Manager Find this file E NORTEL NETWORKS Logs Nnu UTPS 1log If the telephone is properly configured to reach the Business Communications Manager you should see this message in the log Ope
50. and no pre granted ARQ has been issued P0937663 02 0 Chapter 4 VolP Trunk Configuration 83 Business Communications Manager Ottawa send 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 DN421 to the gatekeeper 10 10 10 17 which forwards it to the gateway at 10 10 10 19 and the call 1s established IP Telephony Configuration Guide 84 Chapter4 VolP Trunk Configuration P0937663 02 0 85 Chapter 5 Typical applications This section explains several common installation scenarios and provides examples for how to use VoIP trunks and IP telephony to enhance your network M1 to Business Communications Manager over IP trunks The Business Communications Manager 2 5 can connect with a Meridian to extend an existing network In this type of installation capabilities of the M1 are available to phones on the Business Communications Manager 2 5 For example a company that has a M1 in a Head Office may want to set up a warehouse in another region The warehouse can then call the head office across VoIP lines bypassing long distance tolls Figure 43 M1 to Business Communications Manager network diagram Head Office Warehouse BCM Meridian Telephone PSTN fallback route EN NN Norstar telephone Intranet company Sewer VolP trunk i2004 telephone To set up this system 1 Make
51. ars Figure 37 NetMeeting options Options RAE General Security Audio Video hy directory information E Enter information others can use to find you in the Directory or see while in a meeting with you First name EG Last name Ha E mail address Wo Location Ai ch Comments a Directory Settings Ba Directory uls microsoft com T LI Do not list mp name in the directory W Log onto a directory server when NetMeeting starts Bun NetMeeting in the background when Windows starts W Show the NetMeeting icon on the taskbar Bandwidth Settings Advanced Calling Cancel 3 Click Advanced Settings The advanced settings dialog appears Figure 38 NetMeeting advanced options Advanced Calling Options HE Gatekeeper settings ies TU Use a gatekeeper to place calls 7 Gateway settings m ty M U a gateway to call telephones and videoconferencing dx systems Gateway 47 65 82 1 3l Cancel 4 Under Gateway settings select the Use a gateway option In the Gateway field type the published IP address of the Business Communications Manager 5 Adda remote gateway to your system as explained in Configuring a remote gateway on page 59 When prompted for the IP address of the remote gateway type the IP address of the client computer P0937663 02 0 Chapter 4 VolP Trunk Configuration 77 Repeat this procedure for every NetMeeting client
52. ars on the screen When the greeting appears press the four softkeys located under the telephone s display one at a time from left to right These keys must be pressed one after the other within 1 5 seconds or the telephone will not go into configuration mode If Manual Cfg DHCP 0 no 1 yes appears on the screen you successfully accessed the configuration mode If Manual Cfg DHCP 0 no 1 yes does not appear disconnect then reconnect the power and try to access the configuration mode again For each parameter use the keypad to define values Refer to Table 8 on page 35 for information on each parameter To type a decimal press Table 8 Settings for IP telephones 0 if not using a DHCP server to disperse IP addresses 1 if using 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 on setting up a DHCP server see Configuring DHCP on page 38 SET IP The set IP must be a valid and unused IP address NETMASK This is the subnet mask IP Telephony Configuration Guide 36 Chapter3 Installing IP Telephones DEF GW 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 c
53. ast to coast 1s 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 IP Telephony Configuration Guide 100 Networking 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 1s the result of the following formula serialization delay inms 8 ee mons packepsizeun Dyts link bandwidth in kbfs 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 average IP packet size in bytes EE a p link bandwidth in kbi 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
54. ations Manager Ottawa a caller dialling a set on Business Communications Manager Santa Clara must dial the destination code the destination digits for the Business Communications Manager Santa Clara remote gateway and the DN of the set For example dialing 744322 would connect as follows P0937663 02 0 Chapter 4 VolP Trunk Configuration 75 e 71s the destination code If a suitable level of QoS is available the call is routed through the VoIP trunks e 441s the remote gateway The call is sent across the PDN using the IP address of the Santa Clara Business Communications Manager e 3221s linked to the target line associated with DN 322 The call arrives at the phone with the DN 322 If a user in Santa Clara wanted to make a local call in Ottawa they would dial 7559 followed by the local Ottawa number The digit 7 accesses the VoIP line pool and 55 accesses the Ottawa Business Communications Manager while 9 accesses an Ottawa outside line Connecting an i200X telephone This section takes the example above and uses it to demonstrate how an installer would configure an 12004 phone on the system For information on configuring 1200X telephones see Chapter 3 Installing IP Telephones on page 31 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 ph
55. ator wants to know in advance the amount of traffic on a specific link RA R5 Consider there are four IP telephony ports per site P0937663 02 0 Networking 97 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 from Table 12 on page 95 which is between 24 8 kbit s and 27 6 kbit s peak transmission or approximately 28 kbit s Route R1 R2 needs to support 4 VoIP Calls R4 R5 needs to support eight VoIP calls The incremental peak bandwidth for VoIP traffic 1s therefore RI R2 peak VoIP Load 4 28 kbit s 112kbit s R4 R5 peak VoIP Load 8 28 kbit s 112kbit s With BCM 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 R4 R5 for the four IP telephony ports per site R4 R5 needs to support the incremental bandwidth of 8 x 12 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 Figure 47 Calculating network load with IP telephony traffic Santa Clara i j Santa Clara Richardson traffic Ottawa Tokyo traffic Rickardest Business Communications Manager IP telephony N Router IP Telephony Configuration G
56. atural interruptions that damage communications Packet Group of bits transmitted as a complete package on a packet switched network Packet switched network A telecommunications network based on packet switching technology A link 1s busy for the duration of the packets 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 Voice Compression Method of reducing bandwidth by reducing the number of bits required to transmit voice P0937663 02 0 Index A acronyms 15 C changes to the intranet 102 checklist 27 codecs defined 23 for IP sets 33 Conventions and symbols 14 D Danger symbol 14 default gateway on IP sets 35 DHCP configuring 38 on IP sets 35 Dialing plan 56 E end to end network delay 116 end to end packet los s117 Engineering Specifications 111 F Fallback 123 firmware downloading to IP sets 43 G Gatekeeper defined 22 gatekeeper interoperability 113 H H 323 devices NetMeeting interoperability 111 hop count 100 127 inappropriate load splitting 102 Installation IP sets 31 Interoperability 111 IP telephones 3 port switch 34 Before Installing 33 codecs 33 modifying 40 viewing 36 default gateway 35 defined 19 DHC
57. bers not otherwise assigned will be allowed by the system Click the route you just created In the Use Pool field type the letter of the line pool for the VoIP lines Creating a destination code Creating a destination code allows you to combine the two routes created in Configuring routes on page 63 under a single dialed code When this code 1s dialed the Business Communications Manager will select the VoIP line if possible and if not will fall back to the PSTN line 1 Open Services Telephony Services Call Routing and highlight Destination Codes 2 Click Add The Destination Code dialog appears Figure 31 Add destination code dialog i Property Sheet IBI x Add Destination codes Destination code Save Cancel Ready SESS Warning Applet Window 3 Type a digit for this destination code and click Save 4 Click on the destination code you just created click on the key beside Schedules and click Normal IP Telephony Configuration Guide 66 Chapter4 VoIP Trunk Configuration Figure 32 Normal schedule 47 65 82131 i M amp Normal Add Delete Del A r8 Normal 9 G BCM 4T 55 82 131 amp System 2 Use route Route 000 Resources e Services i Absorbed length an I Telephony Services E System DNs Lines Restriction filters S Call Routing Routes 2 O Destination codes Gr 504 506 e a8 Schedules E3 Mormal 53 Might fe
58. ble 10 on page 93 explain the continuous transmission of a unidirectional media stream 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 See Table 11 on page 94 and Table 12 on page 95 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 1s Numberer calic funr Link bandwidh peak bandwidth per call Number of Calls 2 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 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 103 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 P0937663 02 0 Networking 93 Table 10 Vo
59. compression Conversation Hello Fred This is Susan Do you have a minute Fred here Hi Sure Rx lt gt Bandwidth used Channel Link max Rx channel Bandwidth Time Channel Link max Tx channel Bandwidth Voice frames sent even when speaker is silent When silence compression is enabled voice packets are only sent when a speaker is talking When a voice 1s 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 channel is still equal to the full transmission rate Figure 52 on page 107 below 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 appli cations that can tolerate increased delay and jitter P0937663 02 0 Silence compression 107 Figure 52 One Call on a Full Duplex Link With Silence compression Tx Rx Rx channel Bandwidth Tx channel Bandwidth Conversation Hello Fred This is Susan Do you have a minute lt Fred here Hi Sure Bandwidth used Channel Link max J Time Channel Link max 7 Independent Tx and Rx bandwidth not shared by half duplex calls Bandwidth available for data apps When several calls are made over a full duplex link all calls
60. configuration on page 56 Assign a target line to the DN A target line routes incoming calls to specific telephones DNs depending on the incoming digits This process is independent of the trunk over which the call comes in The mapping of the target lines involves two steps e The incoming digits e g 321 are mapped to a target line e g 241 by setting the Received Number to the incoming digits e g to line 241 to 321 e The target line is mapped to a telephone by assigning the line 241 to the telephone For a detailed explanation of target lines see the Business Communications Manager 2 5 Programming Operations Guide To assign a target line to a DN In Unified Manager open Services Telephony Services System DNs Active Set DNs Choose the DN of the telephone where you want the line to be directed Choose Line Access Line assignment and click the Add button Enter the number of an available target line 241 412 Click the Save button Click on the line number you just created The DN Line screen appears 1n the right frame Ensure that Appearance type is set to Appr amp Ring Oo 0a FP OQO N Go to Services Telephony Services Lines Target Line lt Target line number from step 4 gt P0937663 02 0 Chapter 4 VolP Trunk Configuration 71 8 Click the key beside Trunk line data 9 Click on Received number 10 In the Public number field enter the DN The telephone assigned to that DN can now receive calls route
61. 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 M1 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 VoIP 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
62. d across VoIP trunks Example configuration This section walks through a sample Business Communications Manager configuration In this scenario two Business Communications Managers in different cities are connected to a WAN One Business Communications Manager resides in Ottawa the other in Santa Clara 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 2 5 Keycode Installation Guide For information on Resource Allocation see Configuring the MSC Resources in the Business Communications Manager 2 5 Programming Operations Guide Figure 36 Example PSTN fallback BCM Ottawa BCM Santa Clara Business Communications Manager Ottawa e Private IP address 10 10 4 1 e Public IP address 47 62 54 1 e Phones 221 231 e From this system dial 9 to get onto PSTN Business Communications Manager Santa Clara IP Telephony Configuration Guide 72 Chapter4 VolP Trunk Configuration Private IP address 10 10 5 1 Public IP address 47 62 84 1 Phones 321 331 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 obtains keycodes fo
63. d of the handset cord to the handset 2 Connect one end of a Cat 5 line cordwith RJ45 connectors to the line cord jack on the telephone base Connect the other end of the line cord to the Ethernet connection 3 Plug the AC Power adapter into the base of the telephone and plug the adapter into the AC outlet P0937663 02 0 Chapter 3 Installing IP Telephones 35 Configuring the i2004 telephone Depending on how you set up terminal registration you may be prompted for the password and a DN for the telephone the first time the telephone is booted For information on setting the registration settings see Preparing your Business Communications Manager for IP telephone registration on page 31 1 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 Type the password If you set Auto Assign DN on the Business Communications Manager to OFF the telephone prompts you for a DN Choose a DN for the telephone Entering O will cause the system to automatically assign a DN to the telephone Once registration has been completed you do not need to go through the registration steps described above unless you deregister a terminal To access the local configuration menu on a 12004 telephone 1 Restart the telephone by disconnecting then reconnecting the power After about four seconds the top light flashes and NORTEL NETWORKS appe
64. e For example a caller dials 7 to access the VoIP line 55 to select the remote gateway and then 224 to dial a DN on the remote Business Communications Manager The dialing plan must be set up so that calls are treated the same way by the VoIP and PSTN routes In the example above dialing 755224 must access the same DN even if the call is routed across the PSTN and not through VoIP Activating the VoIP schedule Before activating the VoIP schedule calls using the destination code are routed over the PSTN This 1s because the system is set to use the Normal schedule which routes the call over the PSTN Once the VoIP schedule is activated calls are routed over the VoIP trunk The VoIP line must be activated from the control set for the schedule For information about control sets see the Business Communications Manager 2 5 Programming Operations Guide IP Telephony Configuration Guide 68 Chapter4 VolP Trunk Configuration To activate the VoIP lines 1 Dial Feature 873 from the control set for the VoIP trunk The phone prompts you for a password 2 Type the password Press OK The first schedule appears 3 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 line 1 Dial Feature 873 The phone prompts you for a password 2 Type the password Press OK The system restores Normal schedule Turni
65. e 111 Supported voice payload sizes nannan aaaea 112 Lalaki ih AP eR beh bbe he oe EE eked ABE 113 Quality of voice service llle 116 Site pairs and FOES oua 6484646546 EG da dR EDD OWES EO OES 117 Computed load of voice traffic per link 118 Delay and error statistics 0 0 cee ees 119 11 IP Telephony Configuration Guide 12 P0937663 02 0 13 Preface This guide describes IP Telephony functionality for the Business Communications Manager 2 5 system This includes information on Nortel IP terminals such as the 12004 phone and the 12050 software phone H 323 terminals and H 323 trunks Before you begin This guide is intended for installers and managers of a Business Communications Manager 2 5 system Prior knowledge of IP networks is required Before using this guide you must install and configure a Business Communications Manager 2 5 system This guide assumes e You have planned the telephony and data requirements for your Business Communications Manager 2 5 system e The Business Communications Manager 2 5 is installed and initialized and the hardware is working External lines and terminals and sets are connected to the appropriate media bay modules on the Business Communications Manager 2 5 e Configuration of lines is complete e Operators have a working knowledge of the Windows operating system and graphical user interfaces e Operators who manage the data porti
66. e aggregated for a number of calls Figure 50 on page 105 below 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 50 Two Calls on a Half Duplex Link With Silence compression Tx Rx Tx Rx Chan Bandwidth Conversation oo noches Juan Muy bien y tu Hola Isabella Com o estas Hello Fred This is Susan Do you have a minute lt PR Fred here Hi Sure Bandwidth used Channel Link max pi j Time e put Peak channel bandwidth is n average Bandwidth shared by half duplex calls bandwidth per call 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 Figure 51 on page 106 shows the peak bandwidth requirements for one call on a full duplex link without silence compression Because voice packets are transmitted even when a speaker is silent the peak bandwidth and the average bandwidth used is equal to the full transmission rate for both the transmit and the receive channel IP Telephony Configuration Guide 106 Silence compression Figure 51 One Call on a Full Duplex Link Without Silence
67. eak bandwidth requirement per call on a half duplex link to Peak Bandwidth per call 1 Continuous Transmission Rate peak Band Width per call 1 Continuous Transmission Rate Half Duplex links With Silence Suppression IP Telephony Configuration Guide 94 Networking Table 11 Bandwidth Requirements per BCM Gateway port for half duplex links Codec Payload Type Size Ethernet B W PPP B W FR B W Silence Silence Silence No SP Suppression No SP Suppression No SP Suppression fms koits kbts kbit s kbit s kbit s kbit s kbit s kbit s G 711 2336 233 606 233 65 1952 195 23 195 23 2064 206 43 206 4 64 kb s 180 8 180 8 180 88 161 6 161 65 161 65 167 2 167 23 167 23 163 2 163 23 163 29 150 4 150 43 150 43 154 2 154 23 154 23 8 kb s Baste to so e m us vs laso ws ms uu i 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 Bandwidth Requirements on Full Duplex Links Table 12 on page 95 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 i
68. ed in detail in Jitter Buffer on page 24 To set the jitter buffer size for VoIP trunks 1 InUnified Manager open Services IP Telephony and click on H 323 trunks 2 Click on the Media Parameters tab The Media Parameters dialog appears IP Telephony Configuration Guide 56 Chapter4 VoIP Trunk Configuration Figure 20 Media parameters View Help 47 65 82 131 e Media Parameters Comprehensive e Add Ds i r Media Parameters i BCM 4765821315 System zu Tat Preferred Codec h 29 Y Resources E o Services a 2nd Preferred Codec h 23 Y Telephony Services E IP Telephony 2 3rd Preferred Codec G 71l uLaw Jem on zE 4th Preferred Codec G 7ll aLaw Nortel IP Terminals z H 323 Trunks i Remote Gatewj Jitter Buffer Voice AUTO PortRanges a amp Call Detail Recording LAN CTE Configuration amp Console Service A Voice Mail Voice Button amp DHCP DNS S IP Routing D LANI D WANT D WAN 90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco Web Cache Net Link Mgr 3 In the Voice Jitter Buffer select a value 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 on t
69. eeedan baa hb PANG abu Rt ae haaa FRY 55m 55 ana cal COON O AA 56 Putting VoIP lines into a line pool lle 57 Configuring the Access code for the Line Pool 00000 ees 58 Configuring DNs to access the VoIP line pool 2 0000 eee 58 Configuring a remote gateway 0 ccc eee ees 59 ONG PoTN AAA 60 Setting up a VolP schedule 0 20 61 Conhguring AA 63 Creating a destination code a 65 Configuring IOUS siu su oe PECES VUE EUR cU endte ki od Rm 67 Activating the VOIP schedule iossee reekC er QeE44TTXTERFRPRS TES 67 P0937663 02 0 WA NE nA sedi 405 oe od HAGGANG 0495054 68 Enabling PSTN fallback 0 ccc eee ee 70 Incoming call configuration Lu ue aede dob Re od do deed de ee eod DAA BA 70 Assign a target line tothe DN lllllll Re 10 See a uuenadudasrsemaxa4AmasoARRudur essxbead4cese k Qaeda 71 On Business Communications Manager Ottawa LLL 72 On Business Communications Manager Santa Clara 73 PEN CT EEEE TA Connecting an i200X telephone llle 75 Connecting an i200X telephone on the LAN 0000 00 eee 75 Configuring NetMeeting clients 0 ccc eee eee 75 Quality of Service Monitor 0 aaa 77 duality Ol Service SIBIUS a rus a cet EXERCERE REOR CORRER RR CC C CRCER AR E zT Updating the QoS Monitor data ees 77 igo 9 wor d
70. ent software has any delay measurement modules which can cause a delay distribution measurement for specific site pairs 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 intranet 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 1 3 ms 1 ms lt 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
71. erating area Fax services must not travel on routes that have Fair or Poor QoS levels IP Telephony Configuration Guide 120 Quality of Service If QoS levels of some or all routes fall short of being Good evaluate options and 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 monitor the QoS level reset needs with the end users and respond to user feedback Implementing QoS in IP networks 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 but now need to deliver IP telephony traffic QoS objectives for delay can set an additional design restriction on the intranet One method 1s to subject all intranet traffic to additional QoS restrictions and design the network to the strictest QoS objectives A exact plan to the design improves the quality of data servic
72. es 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 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 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 1s heavy data services can be affected 1f QoS is biased toward IP telephony traffic P0937663 02 0 Quality of Service 121 TCP traffic behavior
73. es Scheduled Services Routing Service and click on VoIP P0937663 02 0 Figure 27 VolP Routing Service 47 65 82 131 Comprehensive 3 YolP r S mg VoIP kak O QU BCM 47 65 82 131 Chapter 4 VolP Trunk Configuration 63 System zz Service setting Manual Resources 244 o Services Ha al Overflow ly I S Telephony Services amp System DNs Lines Restriction filters Call Routing Scheduled Services amp Ringing service nag E amp Restriction service Routing service E Gg Might Evening Lunch VoIP Sched 5 Sched B E amp Common settings System speed dial General settings Hunt groups Companion Hospitality amp Telco features D IP Telephony Call Detail Recording D LAN CTE Configuration amp Console Service 5 Change the Service setting to Manual 6 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 You must configure your PSTN and VoIP line pools before continuing 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 See Configuring digits on page 67 To configure the PSTN route 1 Open Services Telephony Services Call Routing and click on Routes IP Telephony Confi
74. et the QoS monitor to enabled Otherwise set it to disabled For information on enabled QoS see Turning on QoS monitor on page 68 7 Click Save Configuring PSTN fallback By enabling PSTN fallback you allow the system to check the availability of suitable bandwidth for a VoIP call If the bandwidth 1s adequate 1t connects the call across the VoIP line If the bandwidth is not adequate it routes the call across the PSTN 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 the other is a T1 BRI PRI or analog line When a user dials the destination code the system checks first to see if the connection between the two systems can support an appropriate level of QoS If it can the call proceeds as normal If the minimum acceptable level of QoS is not met the call is routed over a different route commonly a PSTN P0937663 02 0 Chapter 4 VolP Trunk Configuration 61 Figure 25 PSTN fallback diagram Before configuring fallback you must have both your VoIP trunk and your fallback line properly configured and line pools created for each For information on creating a VoIP line pool see Putting VoIP lines into a line pool on page 57 You create a PSTN line pool in the same manner Ensure that you use a different line pool for PSTN lines than for VoIP lines Setting up PSTN fallback includes e Setting up a VoI
75. 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 Nortel Networks 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 e 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 from a central point 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 making 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 a LAN frame relay asynchronous transfer mode SONET or wireless connection e Scalability A future proof flexible and safe solution combined with high reliability lets your company focus on customer needs not network problems Nortel Networks Internet Telephony solutions offer hybrid environments that l
76. everage existing investments in Meridian and Norstar systems e Increased customer satisfaction Breakthrough e business applications help deliver the top flight customer service that leads to success By providing your customers rapid access to sales and support personnel via phone the Web and e mail your business can provide better customer service than ever before IP telephones and VoIP trunks This guide describes two similar applications for IP telephony on the Business Communications Manager system IP telephones and VoIP trunks These applications can be used separately or together as a network voice data solution IP Telephony Configuration Guide 20 Chapter1 Introduction 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 Calls made from IP telephones can pass over VoIP trunks or across a Public Switched Telephone Network PSTN VoIP trunks VoIP trunks allow voice signals to travel across IP networks A gateway within the Business Communications Manager converts the voice signal into IP packets which are then transmitted through the IP network The device at the other end reassembles the packets into a voice signal The IP telephony network This section explains the components of the Busi
77. eycode Installation Guide e Business Communications Manager 2 5 Telephone Features Guide IP Telephony Configuration Guide 18 Preface How to get help Your local distributor can provide technical support for your Business Communications Manager system or have access to that information through a Technical Service Center TSC If you require non technical support contact 1 800 4NORTEL 1 800 466 7835 choose option 3 Sales or Pre Sales Support P0937663 02 0 19 Chapter 1 Introduction IP Telephony provides the flexibility affordability and expandability of the Internet to the world of voice communications Business Communications Manager for VoIP gives you 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 the LAN wiring system and eliminates the need for dual cabling Internet Telephony can also eliminate toll charges on site to site calls via global four digit dialing And by using the extra bandwidth on your WAN for IP Telephony 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
78. fer 1 0 eee n 33 Installing and Configuring i2004 Internet Telephones 34 Before Installing Gaede 4 ee eee AA AA OE RI IE D 34 Using a 3 port switch 6 acto 4455 5usd 3 31 409 6 309 9 se OSes KOs eee eae ees 34 Connecting the i2004 Internet telephone 0 0c cece ene 34 Configuring the i2004 telephone cc eee 35 Troubleshooting an IP telephone leeren 36 EDDIOUHDO DIG auueaareedped uerius eeu dreara ER TRSRATEUSS TERR 38 Modifying settings for Nortel IP telephones llle 40 Download Firmware to an i200X telephone 0 000 cece eee eee 43 Deregistering DNs for online IP telephones lesse 44 Deregistering offline DNS ce RII 45 Moving DN ga 6s 644554944 qas vod 33 4 oO SESS pxcddes F4 dud Ex Aaa 46 Configuring the i2050 Software Phone 0 0c cece eee ees 46 FOEDE ooaackaa ream d dci dant dida deb RR RAE ee eee Add ped SG P RS 47 Adding H 323 devices to the system 1 0 ccc ee eee 48 Chapter 4 VoIP Trunk Configuration 0 00 a 51 Installing keycodes 002 aa 51 Setting the Published IP Address 0 a 51 Determining the published IP address 0 000 cee eee 52 Configuring media parameters 0 0 ee eee eee 53 Configuring codecs 2 220240 6455 55054 AREE eae o esse sesesesteusueecass 54 Setting Silence Compression aaa 54 Sander BUNGE abaka Bas
79. gh priority queue m Note Differentiated Service DiffServ is a QoS framework standardized by the Internet Engineering Task Force IETF e nanon 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 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 S witched 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 IP Telephony Configuration Guide 122 Quality of Service The model is the reason for compression characteristics of the codecs Each codec delivers a different MOS for the same network quality Network Monitoring The VoIP Gateway network monitoring function measures the quality of service between the local and all remote gateways on a
80. guration Guide 64 Chapter4 VolP Trunk Configuration Figure 28 Route list 47 55 82 131 Comprehensive Add Del All Gl ACM 4765221315 D System Resources o Services Telephony Services amp System DNs Lines Restriction filters S Call Routing o Routes Route 000 Route 001 Route Route O00 e Destination comes Scheduled Services amp System speed dial amp General settings Hunt groups amp Companion amp Hospitality Telco features D IP Telephony Call Detail Recording D LAN CTE Configuration amp Console Service Voice Mail Voice Button amp DHCP amp DNS Gr IP Routing 2 Click the Add button The Add Routes dialog appears Figure 29 Add route dialog i Property Sheet Add Routes Warming Applet Window 3 Type a number between 001 and 999 to define this route Only numbers not otherwise assigned will be allowed by the system Click the route you just created In the Use Pool box type the letter of the line pool for the fallback lines To configure the VoIP route 1 Open Services Telephony Services Call Routing and click on Routes 2 Click the Add button The Add Routes dialog appears P0937663 02 0 Chapter 4 VolP Trunk Configuration 65 Figure 30 Add route dialog i Property Sheet Add Routes Warming Applet Window 3 Type a number between 001 and 999 to define this route Only num
81. gy is silence compression also known as silence compression or 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 40 of a full duplex conversation is active When one person talks the other listens This is half duplex And 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 50 reduction 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 affects of silence compression on peak bandwidth requirements differ depending on whether the link is half duplex or full duplex Silence compression on Half Duplex Links Figure 48 on page 104 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 IP Telephony Configuration Guide 104 Silence compression Figure 48 One Call on a Half
82. hey 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 Before setting up IP clients you must enable keycodes for IP telephony For information on entering keycodes see the Keycode Installation Guide Preparing your Business Communications Manager for IP telephone registration If this is the first time you are installing an IP telephone on this Business Communications Manager you must activate terminal registration on the Business Communications Manager 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 will connect and boot up without manual registration 1 In Unified Manager open Services IP Telephony and Nortel IP Terminals Select the General tab IP Telephony Configuration Guide 32 Chapter3 Installing IP Telephones Figure 2 Set registration properties Configuration 47 55 82 131 Comprehensive Summary General IP Terminal Status Deregister Offline DH r General Gil BCM 47 55 82 131 D System Resources o Services Telephony Services e IP Telephony System Configuration Nortel IP Terminals Registration ON v Password Auto Assign DNs ON v H 323 Terminals d H 323 Trunks PorRanges
83. his example In the event of link failures spare capacity for rerouting traffic 1s required Some WAN links can exist on top of layer 2 services such as Frame Relay and Asychronous Transfer Mode ATM The router to router link is a virtual circuit which is subject not only toa physical capacity 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 IP Telephony Configuration Guide 92 Networking The difference between the current capacity and its acceptable limit is the available capacity For example a T1 link used at 48 during the peak hour with a planning limit of 85 has an available capacity of approximately 568 kbit s Network Engineering 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 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 Table 10 on page 93 provides bandwidth characteristics for the transmission of voice over IP for various link types given codec type and payload sizes The bandwidths provided in Ta
84. his see Incoming call configuration on page 70 Before telephones on the Business Communications Manager can use VoIP trunks services the VoIP trunks must be placed in a line pool Outgoing call configuration consists of the following Steps e Putting VoIP lines into a line pool e Configuring the access code for the line pool e Configuring DNs to access the VoIP line pool e Configuring a remote gateway e Configuring PSTN fallback P0937663 02 0 Chapter 4 VolP Trunk Configuration 57 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 1s not used for any other type of line To put your lines into a line pool 1 Figure 21 2 Trunk Line data 47 65 82 131 Comprehensive eG Gil BCM 47 55 82 131 D System Resources o Services Telephony Services 7 POPO System DMs Lines OQ VoIP Lines Enabled vol Line 001 g Line 002 Line 003 E amp Line 004 5 Line 001 Trunk line data In Unified Manager open Services Telephony Services Lines VoIP lines Line 001 and click on Trunk Line Data The Trunk Line Data screen appears z Line 001 Trunksine data III S S SS TF JANA aa Line type Pool H I Prime set DN 221 I Distinct rings in u
85. i ons Manager VoIP Corporate intranet Deliver voice Deliver IP fax service lervice 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 IP Telephony Configuration Guide 116 Quality of Service Table 18 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 The settings in Table 18 on page 116 indicate the quality of voice service IP telephony periodically calculates the prevailing QoS level per site pair based on the measurement of the following e one way delay e packet loss e codec 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 1s 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 cu
86. ice over IP Transmission Characteristics for unidirectional continuous media stream Codec IP Type Payload Size Packet Ethernet B W2 PPP B W FR B W L je jeje le jw O 64 kb s m m m mlm v fo fe log la e m la fo jw fe qe C la lab Gas ab G 723 1 24 0 17 6 19 5 6 3 kb s G 723 1 22 9 16 5 18 4 5 3 kb s Notes 1 Gray Background indicated payload sizes used by Business Communications Manager 2 5 for transmission Other values listed indicate payload sizes that the Business Communications Manager 2 5 can receive 2 Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter frame gap Bandwidth Requirements on Half Duplex Links Table 11 on page 94 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 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 1s Peak Bandwidth per call 2 Continuous Transmission Rate Half Duplex links No Silence Suppression 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 p
87. ighest 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 stateof WAN links in the intranet before installing the IP telephony Link utilization This procedure explains how to determine and adjust link utilization 1 Geta 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 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 example for a TI link the threshold 1s higher for example 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 organizations the thresholds can be lower than those used in t
88. ilence compression 104 Two Calls on a Half Duplex Link With Silence compression 105 One Call on a Full Duplex Link Without Silence compression 106 One Call on a Full Duplex Link With Silence compression 107 Two Calls on a Full Duplex Link With Silence compression 108 Relationship between users and services 115 Tables Table 1 Table 2 Table 3 Table 4 Table 5 Table 6 Table 7 Table 8 Table 9 Table 10 Table 11 Table 12 Table 13 Table 14 Table 15 Table 16 Table 17 Table 18 Table 19 Table 20 Table 21 Network diagram cence euce couse soues PAA AA AA 27 Network device checklist llle 28 Network assessment llle hs 28 Resource Assessment llle lees 29 alita so rrr 29 Business Communications Manager system configuration 30 IP telephones IIR G3 30 Settings for IP telephones 0 0 ccc eee 35 OOS SOUS 54 495 04 eh PA ek PAPA ed ek 77 Voice over IP Transmission Characteristics for unidirectional continuous media stream aaan aaa aaaea 93 Bandwidth Requirements per BCM Gateway port for half duplex links 94 Bandwidth Requirements per BCM Gateway port for Full duplex links 95 Link capacity example 64 4 13 3o X4 E ACEOCERAE HECHO EORR OR ED 98 Business Communications Manager 2 5 Product Interoperability Summary 111 Engineering specifications lll
89. ing IP Telephones 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 completed 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 Unistem Trunk Proxy Server UTPS will initiate download as resources become available Deregistering DNs for online IP telephones This command will deregister the selected telephones s from the Business Communications Manager and force it to go through the registration process again Any active calls are dropped To deregister a DN for a phone that is online 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 Figure 10 IP Terminal Status Configuration View SONS AUNE Summary General IP Terminal Status Deregister Offline DN IP Terminal Status 9 tal BCM 47 55 82 131 al DN PANA EEEN EAE amp System 369 Online i2004 47 65 90 26 Default 3002820 Default Resources MET z o Services Telephony Services S IP Telephony system Configuratian Mortel IP Terminals H 323 Terminal
90. ion Guide 34 Chapter3 Installing IP Telephones Installing and Configuring 12004 Internet Telephones The telephone can be configured by the end user or by the administrator If the end user is configuring the telephone the administrator must provide the user with parameters to configure A maximum of 90 IP telephones softphones and H 323 units can be connected on the Business Communications Manager system Before Installing Before installing the 12004 telephone ensure that e the installation site has a 120V AC 60 Hz outlet e the installation site has a 10 100 BaseT Ethernet connection e Tf you are not using the 3 port switch you must have an additional 10 100 BaseT Ethernet connection connecting you to the local Ethernet 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 Using the Nortel Networks 3 port switch the 12004 phone can be installed on the network along with the computer without needing to add an extra cable to a hub For more information consult the 12004 documentation Connecting the i2004 Internet telephone Caution Do not plug the telephone into an ISDN connection This can cause severe damage to the telephone Plug the telephone into only a 10 100 BaseT Ethernet connection To connect the 12004 Internet telephone 1 Connect one end of the handset cord to the handset jack on the telephone base Connect the other en
91. lay and packet loss Repeat this for each site pair At the end of the measurements the results are as shown in Table 21 on page 119 Table 21 Delay and error statistics Measured one way delay Measured packet loss ms Expected QoS level Destination Santa Clara 171 Good Richardson Santa Clara Ottawa Richardson Ottawa Richardson Tokyo Santa Clara mM Tokyo pi 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 1s tight and the IP telephony traffic 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 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 Table 21 on page 119 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 op
92. le calls will still go through but some transfer scenarios will fail P0937663 02 0 Interoperability 113 Gatekeeper The Business Communications Manager is designed to interoperate with any H 323v2 gatekeeper with the Business Communications Manager supporting both Direct GatekeeperResolved and Routed GatekeeperRouted call signalling in this mode of operation Note that if the call signalling method is changed the Business Communications Manager must be restarted before it functions properly Asymmetrical Media Channel Negotiation By default the Business Communications Manager IP Telephony gateway supports the G 729 codec family G 723 1 G 711 u Iaw 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 losses 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
93. lity Summary Vendor mma Tea Nortel Networks Business Communications 2 5 2 0 Manager Nortel Networks i2004 3002B20 or greater Business Communications Manager IP Telephony also interoperates with any H 323v1 or H 323v2 compliant gateway that conforms to the specifications in the following tables Table 15 Engineering specifications Voice compression G 723 1 MP MLQ 6 3 kbit s or ACELP 5 3 kbit s 3 729 CS ACELP 8 kbit s supports plain Annex A and Annex B 3 711 PCM 64 kbit s u A law Silence compression G 723 1 Annex A G 729 Annex B Echo cancellation 48 ms tail delay In band signalling DTMF TIA 464B Call progress Speech path setup methods Call Initiator e H 323 slowStart Call Terminator e H 323 slowStart e H 323v2 fastStart End to end DTMF signaling digits 0 9 and fixed duration tones only IP Telephony Configuration Guide 112 Interoperability Table 16 Supported voice payload sizes Receive transmit to M1 ITG Receive transmit to others G 711 Highest supported by both ends up to 30 20 ms ms in 10 ms increments G 729 Highest supported by both ends up to 30 20 ms ms in 10 ms increments Speech Path Setup Methods Business Communications Manager 2 5 currently only initiates calls using H 323 slowStart methods The Business Communications Manager will however accept and set up calls that have been initiated by another endpoint using H 323v2 fastStart methods as well as H 323 slowStart me
94. m is set up properly Some questions do not apply to all installations 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 VoIP functionality you must have a network diagram that captures all of the information described below If you are configuring IP telephones but not IP trunks you do not need to answer 1 d and 1 e Table 1 Network diagram Prerequisites 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 1 c Does the network diagram contain IP Addresses and network locations of all BCMs 1 d Answer this if your system will use IP trunks otherwise leave it blank Does the network diagram contain IP Addresses of any other Voice over IP gateways desired to connect to 1 e Answer this only if your system will use a gatekeeper otherwise leave it blank Does the network diagram contain alias for any Gatekeeper that may be used Network devices This section of the checklist contains questions about devices on the network such as firewalls NAT devices and DHCP servers IP Telephony Configuration Guide 28 Chapter2 Prerequisites checklist Table 2 Network device checklist Pew eM 2 a This section of the checklist contains questions about devices on the network such as firewalls NAT de
95. mmunications 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 the ports specified for VoIP are allowed An 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 To modify these settings 1 In Unified Manager open Services IP Telephony Port Ranges Figure 39 Port Ranges Configuration nce Wew Help 47 65 82131 i iius Porthanges idd z PortRanges tal BCM 47 55 82 131 PartFianae Ril System ar Resources i o Services Telephony Services S IP Telephony System Configuration Nortel IP Terminals H 323 Terminals H 323 Trunks PortRanges Call Detail Recording LAN CTE Configuration Console Service Voice Mail Voice Button amp DHCP amp DNS IP Routing amp SNMP 205 Monitor amp Web Cache Net Link Mgr Alarm Service amp NAT D OVEN Policy Management NTP Client Settings D a Management amp Diagnostics 2 Select the Port Range you want to modify 3 From the Configure menu select Modify PortRanges The Modify PortRanges dialog appears P
96. nding voice packet latency delay for the Business Communications Manager system IP telephones are e None e Small 06 seconds e Medium 12 seconds e Large 18 seconds QoS routing When it sends a voice frame onto the network the IP telephone firmware places some header information on the frame The header contains the network address of the sending and receiving IP telephones and a TOS Type Of Service byte which contains a routing priority The IP telephone firmware establishes the TOS byte to the highest possible priority so that as the voice frame travels through the network the routers it encounters give it higher routing priority than competing data frames resulting from file transfers WEB downloads e mails etc This process of prioritizing data frames is Quality of Service QoS routing P0937663 02 0 Chapter 1 Introduction 25 The Business Communications Manager system does 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 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 26 Chapter1 Introduction P0937663 02 0 27 Chapter 2 Prerequisites checklist Before you set up VoIP trunks or IP telephones on a Business Communications Manager complete the following checklist to ensure that the syste
97. ndividual transmit and receive channels For instance a 64 kbits full duplex link e g a DSO on TI link has 64 kbits in the transmit direction and 64 kbits in the receive direction 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 P0937663 02 0 Networking 95 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 Table 12 Bandwidth Requirements per BCM Gateway port for Full duplex links Type Size Ethernet B W PPP B W Silence MA Silence PA Silence Suppression Suppression Suppression kbit s kbit s kbit s kbit s 116 8 97 63 103 2 103 23 204 60 8 S6 sse 20 T 752 ma e 60 8 208 ma 236 G 729 8 kb s 0 30 G 723 1 30 12 0 17 6 17 6 6 3 kb s G 723 1 30 22 22 9 11 5 16 5 16 5 5 3 kb s 1 Gray background indicates payload sizes used by Business Communicatio
98. needs IP Telephony Configuration Guide 88 Chapter5 Typical applications Figure 45 M1 to Business Communications Manager network diagram S LUL NS 3 d CS S la SS A 3 ES z SS Jy gs 4 S S 9077 P S7 y BS SI p Pa 7 Y S IG a 47 7 DT y PPL g d 2 S SY NG p SR A S S 2 2 st S S P 8 b S S SS L ZK 8 f X LE amp p LOD s S a 4 d A S wor d 19 B KI LY p A YA IL Vu IL Sons PW amp amp 4 V KL v Cd A zu d 14G a S S PSTN fallback route Intranet VoIP trunk Branch Offices i2004 telephone telephone To set up this system 1 Ensure that the existing network can support the additional VoIP traffic Coordinate a dialing plan On each Business Communications Manager 2 5 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 4 Reboot each system Set up a Call Center on the central Business Communications Manager P0937663 02 0 Chapter5 Typical applications 89 Business Communications Manager to IP Telephones This system allows for home based users or Call Center agents to use the full capabilities of the Business Communications Manager including system users applications and PSTN connections This system does not require VoIP trunk configuration This system functions in a simila
99. ness Communications Manager system and the devices it interoperates with to create a network Figure 1 Network diagram on page 21 shows components of the Business Communications Manager system P0937663 02 0 Chapter 1 Introduction 21 Figure 1 Network diagram 12050 set A a Router TI Md Et eec mc JL Gatekeeper M1 ITG B Business Communications Manager The Business Communications Manager is a key building block in creating your network It interoperates with many devices including the M1 and any H 323 device In this network diagram the Business Communications Manager system is connected to devices through multiple IP networks as well as the PSTN Multiple Business Communications Manager systems can be linked together on a network In Figure 1 Network diagram on page 21 BCM A is connected to a LAN via a LAN card a WAN via a WAN card and a PSTN via Media Bay Modules Through these networks the system accesses other systems and network equipment connected to the network IP Telephony Configuration Guide 22 Chapter Introduction M1 ITG The Meridian 1 Internet Telephony Gateway lets the M1 communicate with H 323 devices including the Business Communications Manager In Figure 1 Network diagram on page 21 telephones on the M1 such as Meridian set A can initiate and receive calls with the other telephones on the system across IP netw
100. ng 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 P0937663 02 0 Chapter 4 VolP Trunk Configuration 69 Figure 34 Remote Gateway list Configuration Performance iu Repo logol Yew Help 47 65 82 131 Comprehensive 2 Remote Gateway y Remote Gateway O QU BCM 47 65 82 131 i Destination IP Transmit Threshold Receive Thresh Gatewa x System 24 020 47628572 Disabled 0 0 0 0 Voice Net Resources 23 1 EN MENS RENS m a o Services Gr Telephony Services S IP Telephony System Configuration Nortel IP Terminals H 323 Terminals S H 323 Trunks Remote Gateway Gr PortRanges Call Detail Recording Gr LAN CTE Configuration Console Service Voice Mail Voice Button amp DHCP amp DNS IP Routing amp SNMP Q05 Monitor amp Web Cache Net Link Mgr Alarm Service 2 NAT D VPN Policy Management NTP Client Settings D a Management 2 Select the Remote Gateway for which you want to enable QoS Monitoring 3 From the Configuration menu click Modify Entry Figure 35 Remote Gateway dialog Property Sheet IDE x Remote Gateway Destination IF 47 62 85 72 Na Re PANA QoS Monitor Enabled Transmit
101. ning signaling channel for set index X at A B C D where A B C D is the telephone s IP address If this entry is not present the IP telephone is not connected to the Business Communications Manager Double check the telephone configuration parameters VNC into the Business Communications Manager open a DOS window and try pinging the telephone Check the configuration settings of any NAT server DHCP server firewall and routers between the telephone and the Business Communications Manager For information on using Ping see Appendix C Network Performance Utilities on page 109 e While signaling between the IP telephones and the Business Communications Manager uses Business Communications Manager port 7000 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 e Ifthe LAN traffic in your network environment is heavy you may experience dropped voice packets If this occurs connect the Business Communications Manager and the telephones to a local network hub to avoid the network traffic e If an IP telephone does not display the text Connecting to server within two minutes after power up the telephone was unable to establish communications with the Business Communications Manager Double check the telephone s IP configurati
102. nk using PPP The preferred codec 1s G 729 kbit s which uses a voice payload of 20 ms Silence compression 1s enabled Given the above what is the peak traffic in kbit s that IP telephony will put on the WAN From Table 12 on page 95 the peak transmission rate for G 729 is 24 8 kbit s per call or 99 2 kbit sin each direction for all 4 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 From Table 12 on page 95 the average bandwidth for each call is 12 4 kbit sec per channel or 49 4 kbit s for all 4 calls for each channel Low priority data applications can make use of bandwidth made available by silence suppression Additional feature configuration This section contains additional information on configuring your network to run efficiently Setting Non linear processing Non linear processing should normally be enabled To set non linear processing 1 In Unified 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 has enough information to load the IP telephony traffic on the intranet Consider the intranet has the topology as shown Figure 47 on page 97 and the installer or administr
103. ns Manager 2 5 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 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 payload of 20 ms Silence compression 1s enabled The Ethernet LAN 1s 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 Table 11 on page 94 the peak transmission bandwidth for G 729 with silence suppression enabled on a half duplex link 1s 34 4 kbit s per call or 137 6 kbit s for all four calls IP Telephony Configuration Guide 96 Networking WAN engineering Wide Area Network WAN links are typically full duplex links both talk and listen traffic use separate channels For example a TI 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 duplexW AN li
104. 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 outside 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 Measurement procedure The following procedure is an example of how to get delay and error statistics for a specific site pair during peak hours P0937663 02 0 Quality of Service 119 Program a script to run the Ping program during intranet s 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 Have the raw output of the Ping results stored in a file Determine the average and standard deviation of one way de
105. on and IP connectivity to the Business Communications Manager cables hubs etc e 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 operational you can increase the contrast level by pressing Feature 7 at the telephone e If the connection between the IP client and the Business Communications Manager is slow ISDN dialup modem change the preferred CODEC for the telephone from G 711 to G 729 See Settings for IP telephones on page 35 IP Telephony Configuration Guide 38 Chapter 3 Installing IP Telephones Configuring DHCP An alternative to manually configuring the IP sets is to use Distributed Host Control Protocol DHCP to automatically assign IP addresses to the IP sets Before setting up DHCP using the information below see the Business Communications Manager 2 5 Programming Operations Guide for more extensive information on DHCP Note Do not enable DHCP on the BCM if you have another DHCP server on the network To set up DHCP to work with IP terminals ensure that DHCP 1s set up with the following settings 1 In Unified Manager navigate to Services DHCP and select the Global Options tab The Global Options screen appears 2 Set the NORTEL IP Terminal Information box to Nortel i2004 A iii jjj kkk 111 7000 1 250 iii jjj kkk 111 7000 1 250 string 1 250 In Figure 3 47 65
106. on consider the impact of incremental IP telephony traffic on routers and LAN resources in the intranet 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 1s 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 telephony traffic from bursty LAN traffic and makes easier the end to end Quality of Service engineering for packet delay jitter and packet loss Getting the best results from your system Further network analysis This section describes how to examine the sources of delay and error in the intranet This section discusses several methods for reducing one way delay and packet loss The key methods are e Reduce link delay on page 100 e Reducing hop count on page 100 e Adjust the jitter buffer size on page 101 Components of delay End to end delay is the result of many delay components The major components of delay are as follows e 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 1s under 18 ms Within the U S the propagation delay from co
107. on of the system are familiar with network management and applications Refer to Chapter 2 Prerequisites checklist on page 27 for more information IP Telephony Configuration Guide 14 Preface 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 A N Alerts you to conditions where you can cause the system to fail or work improperly Note Alerts you to important information Tip 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 1s ping lt ip address gt you enter pang 192 32 10 12 bold Courier text Represent command names options and text that you need to enter Example Use the dinfo command Example Enter show ip alerts routes P0937663 02 0 Preface 15 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 sho
108. one 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 Theinstaller sets up the Business Communications Manager to handle 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 SI IP address 47 62 84 1 This is the published IP address of the Business Communications Manager 3 The Business Communications Manager automatically assigns the telephone the DN of 348 4 The installer sets up a target line for DN 348 using the Received Digits 348 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 321 to connect with telephone 321 dial 9 to access the PSTN or dial 755241 to access a telephone on the Ottawa system Configuring NetMeeting clients NetMeeting is an application available from Microsoft which uses the H 323 protocol IP Telephony Configuration Guide 76 Chapter4 VolP Trunk Configuration To use NetMeeting 1 Install NetMeeting on the client computer 2 In the Tools menu click Options The options dialog appe
109. onnection 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 Setting the Published IP Address on page 51 S1 IP This is the Published IP address of the Business Communications Manager 1 PORT Set this to 7000 1 ACTION Set this to 1 S1 RETRY COUNT Set this to 255 S2 IP This is the Published IP address of the Business Communications Manager S2 PORT Set this to 7000 S2 ACTION Set this to 1 S2 RETRY COUNT Set this to 255 After you have configured a telephone it 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 The IP telephone is completely booted once the date time string 1s displayed on the telephone Do not attempt to use the telephone while the string Connecting to server is displayed initialization of the telephone may take several minutes Troubleshooting an IP telephone If the system is not properly configured several messages can appear e SERVER NO PORTS LEFT The Business Communications Manager has run out of ports This mess
110. 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 voice quality between IP telephones can vary significantly from call to call and time of day When two IP telephones are on a call each IP telephone encodes the speech at the handset microphone into small data packets called frames and 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 Codecs The algorithm used to compress and decompress voice 1s 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 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 1s due to heavy network traffic you can get better voice quality by configuring the IP telephone to use a G 729 CODEC IP Telephony Configuration Guide 24 Chapter1 Introduction Jitter Buffer Voice frames are transmitted at a fixed rate because the time interval between frames 1s constant If the frames arrive at the o
111. orks Telephones The Business Communications Manager system can communicate using several different types of telephones ranging from traditional analog and digital telephones to IP telephones and H 323 terminals Traditional telephones While traditional telephones cannot be connected to the Business Communications Manager system with an IP connection they can useVoIP trunks to make calls to other locations on a network IP telephones Nortel IP telephones include the 12004 12002 and 12050 These devices connect to the Business Communications Manager across an IP network through either a LAN or a WAN H 323 terminals H 323 terminals include computers that have Microsoft NetMeeting installed or other third party devices These terminals connect through the network to either the LAN or WAN card on the Business Communications Manager Gatekeeper A gatekeeper tracks IP addresses of specified devices and provides authorization for making and accepting calls for these devices A gatekeeper is not required for the Business Communications Manager system but can be useful on networks with a large number of devices In Figure 1 Network diagram on page 21 for example when Norstar set A wants to call an H 323 device and Norstar set B is under the control of the gatekeeper Norstar set A sends a request to the gatekeeper The gatekeeper depending on how it is programmed provides Norstar set A with the information it needs to contact Nor
112. quires that each range meet the following conditions Fach 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 IP Telephony Configuration Guide 80 Chapter4 VolP Trunk Configuration Using a gatekeeper 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 services such as e address translation e admission control ARQ e bandwidth control zone management H 323 endpoints 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 alias to IP address translation services Later if an endpoint s IP address changes that endpoint has to re register 1t with the gatekeeper See the gatekeeper software documentation for information about changing IP addresses The call signalling method The call signalling method defines how the Business Communications Manager prefers call signalling information to be directed Call signaling establishes and disconnects a call The Business Communications Manager can use three types of call signalling e Direct Under the direct call signalling method call signalling information is passed directly between endpoints The remote gateway
113. r eight VoIP lines and inputs the keycodes into the system Each Business Communications Manager has 10 phones VoIP lines are available for eight phones but if all 10 phones try to place or receive VoIP calls at the same time two of the calls will be rerouted to the PSTN The installer sets up 221 as the Control set for each line so that the VoIP schedule can be manually activated This setup is necessary for PSTN fallback The installer sets the published IP address Because the public IP network is connected to the LAN 2 connection 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 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 DNs 221 to 231 the installer gives the DN access to line pool O The installer sets up a remote
114. r mannger to the system described in Multi location chain with call center on page 87 This system is less expensive and on a smaller scale However it does not offer PSTN fallback Figure 46 M1 to Business Communications Manager network diagram Central Office Intranet VoIP trunk i2050 Software Phone Home based users E or Call Center agents telephone To set up this system 1 Ensure that each remote user has a network connection capable of support VoIP traffic such as DSL or cable On the Business Communications Manager set up the system to support IP telephones At the remote user s location install and configure 120X X telephone Register each phone and provide it with a DN IP Telephony Configuration Guide 90 Chapter5 Typical applications P0937663 02 0 91 Appendix A Networking This appendix provides information on making your network run more efficiently Determining the bandwidth requirements The design process starts with the an IP telephony bandwidth forecast The bandwidth forecast determines the following 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 h
115. r public or private side Set the network interface on the Private public sideas the published IP address Set the network interface on the private side as the published IP address If you using 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 12004 telephone on page 35 Configuring media parameters There are three steps to configuring media parameters e Configuring codecs e Setting Silence Compression e Setting Jitter Buffers IP Telephony Configuration Guide 54 Chapter4 VoIP Trunk Configuration Configuring codecs This section explains how to select the codecs that are used for VoIP trunks For an explanation of codecs see Codecs on page 23 To configure the codecs 1 InUnified Manager open Services IP Telephony and click on H 323 trunks 2 Click on the Media Parameters tab The Media Parameters dialog appears Figure 18 Media parameters View Help 47 55 82 131 LampieBansiye Local Gateway IP Interface Media Parameters Media Parameters a BCh 47 55 82 131 System zu 1st Preferred Codec G 729 Resources o Services a 2nd Preferred Codec G 723 Telephony Services EE IP Telephony lol 3rd Preferred Codec G 7ll uLlaw
116. ram 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 Sniffer Sniffer is not provided with the Business 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 IP Telephony Configuration Guide 110 Network Performance Utilities P0937663 02 0 111 Appendix D Interoperability Business Communications Manager 2 5 IP Telephony adheres with the ITU T H 323v2 standards and is compatible with any H 323v1 or H 323v2 endpoints 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 12004 telephone and 12050 Software Phone and with the Symbol NetVision IP Phones Table 14 summarizes this information Table 14 Business Communications Manager 2 5 Product Interoperabi
117. re detailed information about this process In order to make calls between Symbol telephones and Business Communications Manager 2 5 each must be configured to have at least one common codec The following codecs are supported by the Symbol telephones e G 711 u law e G 711 A law e G 729 Annex A and Annex B P0937663 02 0 115 Appendix E Quality of Service Setting QoS 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 There are two interfaces that must be considered e IP telephony interfaces with the end users voice services made available need to meet user QoS objectives 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 Figure 54 Relationship between users and services Delay variation Business Communications Manager IP telephony parameters Fallback threshold Silence compression Echo cancellation Non linear programming Business Communicat
118. rrent state of the intranet These measurements help to set accurate QoS needs when using the corporate intranet to carry voice services 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 return to the originating IP address Ping then displays statistics for the packets High packet times can indicate 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 P0937663 02 0 Quality of Service 117 The file can be analyzed by the administrator using spreadsheets and other statistics packages The installer can check if the intranet network managem
119. s H 323 Trunks Gr PorRanges Call Detail Recording D LAN CTE Configuration Console Service Voice Mail Voice Button amp DHCP amp DNS Gr IP Routing amp SNMP Q05 Monitor amp Web Cache Cr Net Link Mgr Alarm Service amp NAT D VPN Policy Management NTP Client Settings D Management Diagnostics 3 Select the IP Terminal with the DN you want to deregister P0937663 02 0 Chapter 3 Installing IP Telephones 45 4 Open the Configuration menu or right click anywhere on the listing for the terminal to bring up the menu 5 Click Deregister DN The Deregister DN option deregisters the selected telephone from the Business Communications Manager and forces it to go through the registration process again It will cause any active call to be dropped Deregistering offline DNs To deregister the DN of an IP client that is offline 1 In Unified Manager open Services IP Telephony and click on Nortel IP Terminals The IP Terminal summary appears 2 Click the Deregister Offline DN tab The Deregister Offline DN screen appears Figure 11 Deregister Offline DN 3 Configuration lepo Hu uu View Summary General IP Terminal Status Deregister Offline DN Deregister Offline DN 47 65 82 131 Comprehensive Qi ACM 4765221315 D System Resources o Services D Telephony Services S IP Telephony
120. se HE RR Distinct ring None I Auto privacy Iy I Use auxiliary ringer IN I Full autahald IN I AllVolP Lines amp Physical Lines amp Target Lines All Lines Restriction filters Call Routing Scheduled Services System speed dial General settings Hunt groups Companion Hospitality amp Telco features Read Only Field In the Line type field set a line pool that is not used by any non VOoIP 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 IP Telephony Configuration Guide 58 Chapter4 VoIP Trunk Configuration Configuring the Access code for the Line Pool An access code is a number that users dial from their telephones to access the line pool containing VoIP trunks Note Set up an access code for the line pool only if you are NOT planning to use PSTN fallback To set up access codes In Unified Manager open Telephony Services General Settings Access Codes Line Pool Codes and click on the line pool that you selected as the VoIP line pool The Pool screen appears 1 Figure 22 Line pool 47 65 82 131 Comprehensive 3 Pool H Jel r Pool H System snood dial o General settings Feature settings 2 EM Timers d Direct dial CAP assignment 8 Dialing plan S Access codes Carrier codes Remote access pi amp COS passwords amp DN lengths
121. 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 Figure 2 6 below 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 available by silence compression is available for lower priority data applications that can tolerate increased delay and jitter IP Telephony Configuration Guide 108 Silence compression Figure 53 Two Calls on a Full Duplex Link With Silence compression Conversation Buenos noches Juan Muy bien y tu ila lt gt BE Rx Hola Isabella Com o estas Tx Hello Fred This is Susan Do you have a minute Rx 5 Fred here Hi Sure Bandwidth used see l Channel Link max Q iz ee Be at xa Xx PA NAR E o2 x O tc cn Time Y Channel Link max oc c5 F og kai Time Peak channel bandwidth is n peak Independent Tx and Rx bandwidth not bandwidth per call shared by half duplex calls Comfort Noise To provide a more natural sound during periods of silence comfort noise is added at the destination gateway when silence
122. so that calls will still be processed correctly if routed over PSTN For information about configuring dialing plans see the Business Communications Manager 2 5 Programming Operations Guide From the control set 221 the installer dials Feature 873 and selects the VoIP schedule VoIP 1s now activated At this point the system 1s configured to make outgoing calls but it 1s not set up to receive incoming calls The installer programs target line 241 with the received digits 221 then assigns this target line to DN 221 This means that calls received containing the digits 221 are delivered to DN 221 The installer repeats this procedure for each DN using different received digits and target lines 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 On Business Communications Manager Santa Clara This procedure details actions that the installer performs to set up the Business Communications Manager Santa Clara 1 The installer obtains keycodes for eight VoIP lines and inputs the keycodes into the system The installer sets up 321 as the Control set for each line so that the VoIP route can be manually activated The installer sets the published IP address Because the public data network PDN is connected to the LAN 2 connection the installer sets the published IP address to LAN
123. star set B IP Network In the network shown in Figure 1 Network diagram on page 21 several LANs and a WAN are shown When planning your network be sure to consider your other needs for a data network P0937663 02 0 Chapter 1 Introduction 23 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 a WAN is any IP network connected to a WAN card on the Business Communications Manager system This may also be a direct connection to another Business Communications Manager system LAN A Local Area Network LAN is a communications network that serves users within a confined geographical area For Business Communications Manager a LAN is any IP network connected to a LAN card on the Business Communications Manager system Often the LAN can include a router that forms a connection to the Internet Public Switched Telephone Network The Public Switched Telephone Network PSTN can play an import role in IP Telephony communications In many installations the PSTN forms a fallback route so that if a call across a VoIP trunk does not have adequate voice quality the call is routed across the PSTN instead The Business Communications Manager also serves as a gateway to the PSTN for users on the system Key IP Telephony Concepts In traditional telephony the voice path between two telephones is circuit switched This means that the analog
124. t the range of IP addresses the DHCP server dispenses 7 Repeat Step 6 for every network adapter that IP sets use to connect to the Business Communications Manager 8 Restart all of the IP telephones Note Whenever changes are made to the DHCP settings telephones will retain gt 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 You must then correct the DHCP settings and restart the telephones Modifying settings for Nortel IP telephones 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 P0937663 02 0 Chapter 3 Installing IP Telephones 41 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 Every IP telephone currently connected to the Business Communications Manager occupies a row in the IP Terminal Status table Figure 6 IP Terminal status Configuration 47 65 82 131 Comprehensive el A IP Terminal Status Gl BCM 47 55 82 131 System 6g Resources i o Services Telephony Services S IP Telephony System Configuration Mortel IP Terminals H 323 Terminals H 323 Trunks PorRanges Call
125. table in the Unified Manager contains a mapping of phone numbers which the Business Communications Manager uses to perform DN to IP address resolution e Gatekeeper Resolved Gatekeeper Resolved signalling uses a gatekeeper for call permission and address resolution All call signalling occurs directly between H 323 endpoints In effect the gateway requests that the gatekeeper resolves the phone numbers into IP addresses but the gatekeeper is not involved in call signalling e Gatekeeper Routed Gatekeeper Routed signalling uses a gatekeeper for call permission and address resolution In this method call signalling is directed through the gatekeeper For information on changing the call signalling method see Alias names on page 80 Note 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 signalling method to use Alias names 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 P0937663 02 0 Chapter 4 VolP Trunk Configuration 81 E 164 numeric identifier containing a digit in the range 0 9 commonly used since it fits into dialing plans Identified by the keyword TEL e H323Identifier alphanumeric strings representing names e mail addresses etc Identified by
126. ther 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 degrades 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 The IP telephone firmware places 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 send 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 The disadvantage of using a jitter buffer 1s that the speech arrives delayed by the number of frames in the buffer For one sided conversations this is not an issue For conversations where one party tries to interrupt the other speaking party it is annoying because 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 correspo
127. thods Note that when using Symbol phones with Business Communications Manager 2 5 the Symbol phones will have to be configured to use H 323 slowStart only due to interoperability issues with fastStart between the Symbol telephones and the Business Communications Manager 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 phone Business Communications Manager 2 5 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 codecs on page 54 If this codec 1s G 729 or G 723 the Silence Suppression option must be the same on all Business Communications Managers involved e If interworking with a Meridian I 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 e Symbol telephones must be configured to use the same codec as the preferred codec on Business Communications Managers in the system If these rules are not adhered to simp
128. tional base units can be added P0937663 02 0 Chapter5 Typical applications 87 Figure 44 Multiple Business Communications Manager systems network diagram Head Office Warehouse orstar Communications ommunications Telephone Manager Manager Telephone 7 PSTN fallback route Intranet VoIP trunk i2050 software phone i2004 telephone In B ccc DC d Remote Office To set up this system Ensure that the existing network can support the additional VoIP traffic Coordinate a dialing plan On each Business Communications Manager 2 5 system e Set up outgoing call configuration for the VoIP gateway e Set up a remote gateway for other Business Communications Managers or NetMeeting users e Set phones to receive incoming calls through target lines e Configure the PSTN fallback and enable QoS on both systems 4 Reboot each system This system uses fallback to PSTN so that if traffic between the Business Communications Manager systems becomes too heavy calls are routed across the PSTN Multi location chain with call center In this installation one Business Communications Manager runs a Call Center and passes calls to the appropriate branch offices each of which have 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 the the remote office best able to handle their
129. uide 98 Networking Enough link capacity Table 13 on page 98 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 13 Link capacity example Incremental IP telephony Available LA gs capacity Traffic Enough LAM x i kbit s capacity R1 R2 Santa Clara Ottawa Santa Clara Tokyo R4 R5 1536 Santa Clara Richardson Ottawa Tokyo Santa Clara Tokyo Some network management systems have network planning modules that determine network flows These modules provide more detailed and accurate analysis as 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 P0937663 02 0 Networking 99 Other intranet resource considerations Bottlenecks caused by non WAN resources do not occur often For a more complete evaluati
130. unks nag Remote Gatewi PortRanges ag amp Call Detail Recording LAN CTE Configuration Console Service aay Voice Mail Voice Button amp DHCP DNS S IP Routing D LANI D WANT D WAN 90 1 ModemBack amp SNMP o QoS Monitor Mean Opinion Sco Web Cache Net Link Mgr iy g A 2 In the Configuration menu click Add DN The H 323 Terminal List dialog appears P0937663 02 0 Chapter 3 Installing IP Telephones 49 Figure 15 H 323 Terminal list dialog Property Sheet DM aaa Password IP Address 10 10 10 Ds R eady Warming Applet Window 3 Type a valid DN password and the IP address of the terminal 4 Click the Save button 5 Connect the H 323 device to the network and use the documentation available with that device to install it IP Telephony Configuration Guide 50 Chapter3 Installing IP Telephones P0937663 02 0 51 Chapter 4 VoIP Trunk Configuration This chapter explains how to configure VoIP trunks on a Business Communications Manager 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 phone on the Business Communications Manager system They are not strictly for use with IP terminals Configuring a VoIP trunk requires the following actions
131. use DHCP The Scope Specific Options Screen appears IP Telephony Configuration Guide 40 Chapter3 Installing IP Telephones Figure 5 DHCP range Configuration Tools View Help Cinnwahoneites Scope Specific Options Address Hange Hxx Excluded Address E xx Reserved Addr _ Scope Specific Options TORE d BCM 47 65 52131 2l amp System al Name a ap DHEPLAN1 Resources 7 Services Description Pana aaa Telephony Services a DHCPLANIT IP Telephony al HBBHBHBHBHH amp Call Detail Recording DNS Server PrimarlP S amp condarylP 06 47 65 82 131 LAN CTE Configuration a Console Service a WINS Server 044 j amp Voice Mail I Voice Button 2 e DHCP Default Gateway 03 Local Scope al LANI Lease Time in secs D51 259200 amp Remote Scope E a e Scope Status Enabled 7 amp SNMP 2 IBHHHHBHBHBHHB ESL BUSELEIEEHELELE ELE BBBBHBHBHBHBHBHBHSSES 205 Monitor amp Web Cache Net Link Mgr Alarm Service amp NAT 2 VPN Policy Management NTP Client Settings D Management amp Diagnostics Format nane 6582130 6 Seteach of the following e Scope Status set this to Enabled e Default Gateway Field set this to Published IP Address e Address Range se
132. vices and DHCP servers 2 b Is the network using private IP addresses If the network uses public IP addresses complete 2 c If the network uses private IP addresses complete 2 d to 2 e 2 c Are there enough public IP addresses to accommodate all IP telephones and the Business Communications Manager 2 d Does the system have a firewall NAT device or will the BCM be used as a firewall NAT device 2 e The Business Communication Manager has limited space fore firewall rules If the Business Communications Manager is to be used as a firewall NAT device do the firewall rules fit within 32 input rules and 32 output rules 2 f 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 This section ensures that the network 1s capable of handling IP Telephony and that existing network services are not adversely affected Table 3 Network assessment LI eM 3 a Has a network assessment been completed pf 3 b Has the number of switch hub ports available and used in the LAN infrastructure been calculated 3 c Have the used and available IP addresses for each LAN segment been calculated Ba DTTIITTT T T pa Ha naman ratio Lon aa Dt as me mater aues beween paraan ce Pa naman a anes ara pi as me aua are way on pa nor nn ca Resource Assessment You must allocate sufficient resources on the Business Communications Manager for IP telephony
133. w at valid routes valid route is one variable and you substitute one value for it plain Courier Represents command syntax and system output such text as prompts and system messages Example Set Trap Monitor Filters Acronyms This guide uses the following acronyms ATM Asynchronous Transfer Mode BCM Business Communications Manager CIR Committed Information Rate DID Direct Inward Dialing DOD Direct Outward Dialing DIBTS Digital In Band Trunk Signalling DSB DIBTS Signalling Buffer ITU International Telecommunication Union IXC IntereXchange Carrier IP Internet Protocol ISDN Integrated Services Digital Network LAN Local Area Network LATA Local Access and Transport Area LEC Local Exchange Carrier MOS Mean Opinion Score PCM Pulse Code Modulation PPP Point to Point Protocol PRI Primary Rate Interface PSTN Public Switched Telephone Network QoS Quality of Service RTP Real time Transfer Protocol SNMP Simple Network Management Protocol TCP Transmission Control Protocol IP Telephony Configuration Guide 16 Preface UDP User Datagram Protocol UTPS UNISTEM Terminal Proxy Server VoIP Voice over Internet Protocol WAN Wide Area Network P0937663 02 0 Preface 17 Related publications Documents referenced in the Business Communications Manager 2 5 IP Telephony Configuration Guide include e Business Communications Manager 2 5 Installation and Maintenance Guide e Business Communications Manager 2 5 Software K
134. ways on one larger and more powerful router 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 Quality decreases with lost packets Increase the size of the jitter buffer to improve quality when jitter is high IP telephony fax calls use a fixed jitter buffer that does not change the hold time over the duration of the call Fax calls are more prone to packet loss In conditions of high jitter increase delay through the use of a deeper jitter buffer To allow for this increase IP telephony provides a separate jitter buffer setting for fax calls 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 e

Avaya BCM 2.5 IP Telephony Configuration Guide

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