Home

Avaya IP Telephony Configuration Guide

1. 2 When the call is received from the public network at System A Santa Clara the system recognizes that the received number is not a local system number The call is received as a public call 3 System A has a route and destination code that recognizes the received number and routing code as belonging to the route that goes to System B Ottawa System A passes the call to System B over a dedicated trunk in this case a VoIP trunk This call is now designated as a private call type P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 111 Dialout 2244 399 jc OO dedicated VoIP trunk private network Ottawa i gt Gateway 3 4 System B recognizes the code as its own and uses a local target line to route the call to the correct telephone Target line XXX recognizes 2244 DN 2244 assigned with target line XXX DN 2244 IP Telephony Configuration Guide 112 Chapter6 Setting up VoIP trunks for outgoing calls Setting up VoIP trunks for fallback Fallback is a feature that allows a call to progress when a VoIP trunk is unavailable or is not providing adequate quality of service QoS Refer to the information under the following headings for details about setting up fallback for VoIP trunks e Des
2. 2 00 ee ee 85 Pre installation system requirements 0 0c eee eee eee ees 86 Pee oo bth oa ee ales onda ah aie a dane al dus erar id 86 Published IP addres9S i cc cet cese cea Rome eRe Oe Ra ew ee x aE 86 SIP network Gate considerationis 1 cuu cusa sa A RRR eR REE DA dE CR LEE 86 H 323 network applications considerations 00000 0c eee eae 86 mneursgia i c ert PR 87 Determining the IP trunk Count issus Ry Rr RR RA E E ER x xn 87 Configuring media parameters sscccseeessoess ee RE RRERX REG ReRE LERYS 89 Seting Up the local DG IG WE duse Bee X uq ade qiiia bong ek idt ia eal deo ub A 91 Modifying local gateway settings for H 323 and SIP trunks 92 Notes about NPI TON aliases for H 323 trunks llle 96 Setting up SIP trunk subdomain names llllllllseees 97 Viewing SIP summary and status ioooseeenusce eem mmc mme mme com ee ams 98 Incoming Galle Assigning target lines 2 usi uaa bg Roca Rab ode qoe x dba tob Ud aod 98 How tonse Target OOo d adco d pedo DEF Rae rade e pq vd dora aibqete rad us 98 Configuring target INOS oso G6 toad cad ESS RP YRPSERERPSexsPRPEE XP 99 Chapter 6 Setting up VoIP trunks for outgoing calls cece ee eee 101 Setting up remote gateways and end points 0200 0c eee eee eee 102 Configuring a remote gateway H 323 trunks 0000 e eee eee 103 Creating a remote gateway record 00 e eee eee 103 Conf
3. 172 Appendix C Network performance utilities leeren 173 Appendix D lnteroperabilii s quaa is ERAPRAARTAARAWERERERAMRARARE ARAARRRETERAA 175 Speech path setup MIPS osse snas ead dere dh Cn ARE CR HR REC ORA 176 Media PAM UIE ari aE A 177 GalakegDOr saci 2a derert GAS s reai iiaii reat Eaki aatia 177 Asymmetrical media channel negotiation Net Meeting 2 00 178 No feedback busy SIAO 24 suse ER XGARWRORRAORRUROR AUR ROR irer daw EUR SHER CR 178 Setting up Remote Routers for IP Telephony Prioritization 179 Creating sn outbound traffic fiter uisus scusa ee riani etaa 179 Sample criteria ranges and actions for UDP filtering 180 Using VLAN on the HelWOIK 2626555 s65 4 oi 4 oo ebb bY on ea 845s Heeb SERENE 181 Choosing DHCP fot VLAN a lt cuks oeasds ees TENET PR ET Gens IOPETQS NS 181 Specifying the site specific options for VLAN 0 0 00 eee eee ee eee 182 Symbol NetVision telephones uiua ssaaaceixa akon ee te kak abe 183 Software interoperability compatibility and constraints 085 183 H 323 trunk compatibility by software version llli 183 H 323 trunk compatibility iS8UBS 2 0 2 2022 40 46 20eie0ues Hed RR ERE X ae 184 SIP trunk interoperability issues cssc c mmm mx 188 T 38 fax restrictions and requirements 0 00 eee eee 189 Appendix E AIR gqg TET C 191 Selling QOS
4. Send Name Display Y N If the remote voice mail system resides on a Meridian 1 system that system should have the MWI package to allow message waiting indicators to occur on network telephones In this case the IP trunking Remote Capability MWI field should be set to Yes the default to indicate that the Business Communications Manager is compatible with the M1 If the M1 does not have the MWI package you need to set the IP trunking Remote Capability MWI field to No to indicate that there is no compatibility Note SIP trunks do not IP Telephony Configuration Guide 144 Chapter 7 Optional VoIP trunk configurations Table 38 IP trunking interoperability fields Continued Field Value Description Remote Capability MWI Y N The public or private OLI outgoing line identification are separately configurable for each telephone under Line Access Therefore when the VoIP trunks allow name display on outgoing calls Send Name Display the system will send the appropriate OLI based on line type Public or Private Default is Y Virtual Private Network ID digits Default 0 This is the VPN ID for a remote system such as Succession 1000 M In some applications such as for the Survivable Remote Gateway SRG acting as a Branch Office this ID is required to ensure that Bandwidth Management is handled correctly for calls coming into the Succession 1000 M from your system Zone ID
5. Field Value Description Name lt alphanumeric gt Enter an indentifying tag for the remote system Destination IP lt ip address gt Enter the IP address of the remote system gateway QoS Monitor Disabled Enabled Choose Enabled if you intend to use a fallback PSTN line Ensure that QoS Monitor is also enabled on the remote system Otherwise choose Disabled For information about QoS refer to Quality of Service Monitor on page 127 Transmit Threshold 0 0 bad to 5 0 excellent Enter the Mean Opinion Score MOS that the system uses to determine when a call needs to fallback to a PSTN line If the MOS on the transmit channel falls below this value for all of the available codecs the BCM will fallback to a PSTN line For more information about MOS refer to Appendix E Quality of Service on page 191 and the QoS Monitor chapter of the Programming Operations Guide P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 105 Table 25 Remote gateway record Continued destination code for the route to this system Field Value Description Receive Threshold 0 0 bad to 5 0 Enter the Mean Opinion Score MOS that the system uses to excellent determine when a call needs to fallback to a PSTN line If the MOS on the receive channel falls below this value for all of the available codecs the BCM will fallback to a PSTN
6. System grabs VoIP line pool and dials out 92045554678 UDP network The user dials 82233 remote system DN 2233 destination digits private access code 555 The system absorbs the 8 but then adds the private access code to the dialout digits If the call falls back to PSTN line the system still only absorbs the 8 then dials out the private access code private network PSTN line or public access number public PSTN to the remote system in front of the 2233 IP Telephony Configuration Guide 122 Chapter6 Setting up VoIP trunks for outgoing calls Example A private network configured for fallback This section walks through a sample Business Communications Manager configuration including e System programming for networking and fallback routes on page 123 e Making calls through a private VoIP network gateway on page 125 e Connecting an i200X telephone on page 125 In this scenario shown in the following figure two Business Communications Managers in different cities are connected through a WAN One Business Communications Manager resides in Ottawa the other resides in Santa Clara Both VoIP trunks and an PRI SL 1 line connect the system in a private network Figure 40 Example PSTN fallback DN 3322 DN 2244 Dialout Dialout IP network aaa Packet Data Network 3322 Ottawa i Santa Clara Gateway 2 Gateway 3 Gateway destination Gateway destination e M
7. e Scope Specific Options tab Scope Status Enabled Default Gateway Field lt Published IP Address gt Address Range tab contains the range of IP addresses you need Restart all existing connected IP telephones Note Whenever changes are made to the DHCP settings telephones will retain the old settings until they are restarted IP Telephony Configuration Guide 54 Chapter 3_ Installing IP telephones If the DHCP server is not properly configured with the Published IP address the telephones will display Invalid Server Address If this message appears correct the DHCP settings and restart the telephones IP telephony DHCP notes Nortel IP telephones supports two forms of DHCP configuration full and partial If partial DHCP is selected the user must manually enter the primary and secondary Business Communications Manager address action retry count The IP telephone then configures a IP address netmask and default IP gateway via DHCP If full DHCP is selected the IP telephone configures all parameters via DHCP Note If partial DHCP is selected the DHCP server does not need to send the vendor specific or site specific information outlined below The information below pertains to Full DHCP only In the case of partial DHCP the IP telephone requires only the Router option and Subnet Mask option to configure along with IP address and lease time Full DHCP support in the IP terminals requires sending a Class Identifier
8. Absorbed length Third Route Absorbed length 1 Change First Route to the route you configured for your VoIP line 2 Set the Absorbed length to absorb the amount of the destination code that is not part of the dialout for the trunk For example If the remote gateway destination digit is 2 which is part of the remote system DN structure CDP network and you specified a destination code of 82 set this field to 1 so that the 2 is still part of the dialout If the destination code is different from the remote gateway destination digits and you entered an External into the route record the destination digit for the remote system set the absorbed length to the number of digits in the destination code The system will dial out the External you entered in front of the rest of the number that the user dialed This would occur if the network is set up with a UDP dialing plan Note Do not add alternative routes second or third Since fallback is active the system immediately falls back to the Normal schedule if the first route is not available IP Telephony Configuration Guide 118 Chapter6 Setting up VoIP trunks for outgoing calls Normal schedule for all fallback destination codes Figure 37 Normal schedule routing information 5A Normal Use route Absorbed length 1 Change Use Route to the route you configured for your PSTN fallback line the line to the other system 2
9. IP Telephony Configuration Guide 160 Efficient Networking 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 in the figure below and the installer or administrator wants to know in advance the amount of traffic on a specific link R4 R5 Figure 56 Calculating network load with IP telephony traffic Santa Clara b 4 Santa Clara Richardson traffic Ottawa Tokyo traffic Minti a ants aa Santa Clara Tokyo traffic Richardson Business Communications Manager IP telephony IN Router P0609327 02 Efficient Networking 161 Each site supports four VoIP ports Assume the codex is G 729 Annex B 20 ms payload Assuming full duplex links peak bandwidths per call are between 24 8 kbit s and 27 6 kbit s peak transmission or approximately 28 kbit s This is shown in the following figure taken from the table under Bandwidth requirements on full duplex links on page 156 Figure 57 Network loading bandwidth PPP B W FR B W Payload Silence Silence Size No SP Suppression No SP Suppression ms peak peak Avg peak peak Avg Codec Type kbit s kbit s kbit s kbit s kbit s kbit s G 729 8 kb s 30 19 2 9 6 21 1 10 6 Route R1 R2 needs to support four VoIP Calls R4 R5 needs to support eight V
10. Table 23 Local Gateway IP interface fields Continued Field Value Description Gateway Protocol None SL1 CSE Both these protocols require a keycode SL1 use this protocol only for BCM 2 5 systems CSE Use this protocol for BCM 3 0 and newer systems This protocol supports Meridian 1 IPT Otherwise use None Gateway Protocol none none SLI CSE H245 Tunneling Disabled Enabled Default Disabled If Enabled the VoIP Gateway tunnels H 245 messages within H 225 The VoIP Gateway service must be restarted for a change to take effect Call Signaling Port 0 655835 Default 1720 This field allows you to set non standard call signaling port for VoIP applications that require special ports 0 The first available port is used Ensure that you do not select a port that has been assigned elsewhere in the Business Communications Manager RAS Port 0 655835 Default 0 This field allows you to set a non standard Registration and Admission RAS port for VoIP applications that require special ports 0 The first available port is used Ensure that you do not select a port that has been assigned elsewhere in the Business Communications Manager These fields are mandatory when you use Radvision ECS 2 1 0 1 These fields are optional when you use Radvision ECS 2 1 0 1 Fields that appear only for SIP trunks SIP Domain lt name gt com Enter
11. Destination Digits 5955 4 Use the information in the table below to set up the gateway information Table 26 Adding SIP Address Book records Field Value Description Name lt alphanumeric gt Enter an indentifying tag for the remote system Destination IP lt ip address gt Enter the IP address of the remote system gateway P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 107 Table 26 Adding SIP Address Book records Continued Field Value Description QoS Monitor Disabled Choose Enabled if you intend to use a fallback PSTN line Enabled Ensure that QoS Monitor is also enabled on the remote system Otherwise choose Disabled For information about QoS refer to Quality of Service Monitor on page 127 Transmit Threshold read only Receive Threshold read only Destination Digits lt numeric gt Set the leading digits which callers can dial to route calls could be the through the SIP trunk Ensure that there are no other same as the destination SIP endpoints currently using this combination of destination code digits for the route to If multiple leading digits map to the same destination this system separate them with a space For example 7 81 9555 These numbers are passed to the remote system as part of the dialed number Outgoing call configuration This section explains how to set up your system to place calls through VoIP trunks The s
12. lt digits gt Default 0 A remote system such as Succession 1000 M may configure your system into a separate zone to accommodate specific dialing requirements such as for an SRG system acting as a Branch Office to a Succession 1000 M system The system administrator of the Succession 1000 M system provides the Zone ID Enter that number here and include it in any destination codes directed to or through that system so that the remote system can correctly direct incoming calls 4 Click anywhere off the IP trunking dialog to save the changes P0609327 02 Chapter 7 Optional VoIP trunk configurations 145 Configuring NetMeeting clients NetMeeting is an application available from Microsoft which uses the H 323 protocol To use NetMeeting 1 Install NetMeeting on the client computer 2 Inthe Tools menu click Options The options dialog box appears Figure 48 NetMeeting options Options Audio 3 Click Advanced Calling The Advanced Calling Options dialog appears IP Telephony Configuration Guide 146 Chapter 7 Optional VoIP trunk configurations Figure 49 NetMeeting Advanced Calling Options Advanced Calling Options 4 Under Gateway settings select the Use a gateway option 5 Inthe Gateway field type the published IP address of the Business Communications Manager 6 Click the OK button 7 Addaremote gateway to your system as explained in Setting up remote gateways and
13. Chapter 5 Configuring local VoIP trunks This section explains how to configure voice over IP VoIP trunks on a Business Communications Manager system for incoming traffic A VoIP trunk allows you to establish communications between a Business Communications Manager and a remote system across an IP network The Business Communications Manager supports two trunk protocols H 323 version 4 and SIP Since these protocols have different properties they are configured through separate records even though many of the same settings are required H 323 trunks support connections to other Business Communications Managers Meridian systems running IPT software and trunk based applications such as NetMeeting SIP trunks currently support trunk connections between Business Communications Managers e H 323 trunks are programmed under Services IP Telephony IP Trunks H 323 Trunks e SIP trunks are programmed under Services IP Telephony IP Trunks SIP Trunks Each trunk is also associated with a line record which are found under Services Telephony Services Lines VoIP lines Configuring a VoIP trunk requires the following actions e Pre installation system requirements on page 86 e Counting IP trunks on page 87 e Configuring media parameters on page 89 e Setting up the local gateway on page 91 e Viewing SIP summary and status on page 98 e Incoming calls Assigning target lines on page 98 Note If you are u
14. G 723 1 5 3 30 22 9 22 9 11 5 16 5 16 5 8 3 18 4 18 4 9 2 kb s Notes 1 indicates payload sizes used by Business Communications Manager 3 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 P0609327 02 Efficient Networking 157 With no silence suppression both the transmit path and the receive path continuously transmit voice packets Enabling silence suppression on full duplex links reduces the average bandwidth However since transmit and receive paths use separate channels the peak bandwidth per call per channel does not change Therefore peak bandwidth requirements per channel Rx or Tx per call on a full duplex link is Peak Bandwidth per channel per call 2 Continuous Transmission Rate Full Duplex links With or Without Silence Suppression The bandwidth made available by silence suppression on full duplex links with continuous transmission rate average bandwidth requirement is available for lower priority data applications that can tolerate increased delay and jitter LAN engineering examples Example 1 LAN engineering voice calls Co
15. H H 323 fallbacksetting 93 gateway specifications 176 non linear processing 159 Trunks record jitter buffers 91 IP Telephony Configuration Guide 214 Index H 323 devices NetMeeting 175 NetVision 73 H 323 endpoints 133 H 323 terminals record deleting handset record 82 NetVision 79 updating 81 H 323 Trunks record 89 remote gateway 103 106 H323Identifier 94 half duplex links bandwidth requirements 155 silence compression example 168 silence suppression 156 handset changing name 82 deleting record 82 home based users 152 hop count reducing 164 hot desking change password 61 Hunt group target line to DN 98 i2001 connecting 125 feature labels 63 keep DN alive 69 server parameters 49 12002 connecting 125 feature labels 63 keep DN alive 69 server parameters 49 12004 connecting 125 feature labels 63 keep DN alive 69 server parameters 49 12050 Software Phone configuring 71 keep DN alive 69 server parameters 49 IEEE Address H 323 terminals list also see ESS ID 81 inappropriate load splitting 165 in band signaling 176 Incoming call configuration 98 incremental IP telephony traffic 162 Installation 3 port switch 46 configuration display keys 48 12050 Software Phone 71 initialization IP telephones 51 IP telephone server parameters 49 IP telephones 41 NetVision telephones 73 NetVision before you start 75 post installation network measurements 165 restart to configure 48 Unified Manage
16. P0609327 02 Efficient Networking 165 Saturation refers to a 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 164 Routing issues Routing problems cause unnecessary delay Some routes are better than other routes The Traceroute program allows the user to detect routing anomalies and to correct these problems Possible high delay differences causes are routing instability e wrong load splitting frequent changes to the intranet e asymmetrical routing Post installation network measurements The network design process is continuous even after implementation of the IP telephony and commissioning of voice services over the network Network changes in regard to real IP telephony traffic general intranet traffic patterns network controls network topology user needs and networking technology can make a design invalid or non compliant with QoS objectives Review designs against prevailing and trended network conditions and traffic patterns every two to three weeks at the start and after that four times a year Ensure that you keep accurate records of settings and any network changes on an ongoing basis Ensure that you have valid processes to monitor analyze and perform design changes to the IP telephony a
17. Also Registration must be turned on in the Services IP Telephony IP Terminals Nortel IP Terminals General page before the telephone can access the system to register Change the contrast level When an IP telephone is connected for the first time the contrast level is set to the default setting of 1 Most users find this value is too low Therefore after the telephone is installed use FEATURE 9 and use the UP or DOWN key to adjust the contrast Block individual IP sets from If you want to block one or more IP telephones from calling outside the dialing outside the system System use Restriction filters and assign them to the telephones you want to block Restriction filters are set up under Services Telephony Services Restriction filters Restriction filters are discussed in the Programming Operations Guide P0609327 02 Chapter 3 Installing IP telephones 53 Configuring DHCP You can use DHCP to automatically assign IP addresses to the IP telephones as an alternative to manually configuring IP addresses for IP telephones If you are using the Business Communications Manager as the DHCP server you can also configure the server to automatically locate the VLAN ID for the system and assign it to the telephones that register Before setting up DHCP using the information in Setting up DHCP to work with IP terminals refer to the Programming Operations Guide for detailed information about DHCP Also refer
18. If you want to specify a range of DNs you can use the Add Users Wizard This wizard is explained in the Programming Operations Guide Caution If your system uses the Call Center application there is a potential conflict for DN assignment if you choose to allow the system to auto assign DNs to your handsets In this case it is recommended that you manually configure the NetVision DNs before allowing them to register to the system DN records for NetVision handsets are created in the same way as for all other telephones on the system The various settings for DN records are described in the Programming Operations Guide Choose model IPWIs IP Wireless when configuring NetVision DN records Once the handset registers with the system the DN also appears under Telephony services System DNs DN Registration IP wireless DNs reg d Active Inactive If you need to deregister the handset you can use the Configuration menu under this heading Deregistering a telephone on page 83 If you need to change the H 323 Terminals record refer to Updating the H 323 terminals record on page 81 and Deleting a NetVision telephone from the system on page 82 If you require information about changing the DN records refer to the Programming Operations Guide for details P0609327 02 Chapter 4 Installing NetVision telephones 79 Adding a NetVision record in the Unified Manager Follow these steps to preconfigure an H 323 Ter
19. To facilitate this system you need to ensure that the routing codes on the non PSTN system point to the system connected to the PSTN and then to the PSTN On the PSTN connected system the system and routing codes must be configured to recognize and pass public calls from the other system out into the PSTN network Since the receiving PSTN sees the calls as remote dial ins ensure that the correct remote access packages have been established for the VoIP trunks This also means that if the VoIP trunks are inaccessible between the systems there is no provision for a fallback route Figure 52 Routing all public calls through one Business Communications Manager Head Office Warehouse Business Business System Communications Communications telephone Manager Manager System telephone Intranet VoIP trunk i2050 Software i2004 telephone telephone remote i2004 The Programming Operations Guide provides a detailed description of the configurations required for tandeming a system over PRI lines Except for the VoIP trunk requirements the system and routing configurations would be similar P0609327 02 Chapter 8 Typical network applications using MCDN 151 Multi location chain with call center You can create a multi location chain where one Business Communications Manager runs a Call Center and passes calls to the appropriate branch offic
20. and varying arrival rates of consecutive voice packets The greater the size of the jitter buffer the better sounding the received voice appears to be However voice latency delay also increases Latency is very problematic for telephone calls as it increases the time between when one user speaks and when the user at the other end hears the voice Note You can only change the jitter buffer on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone Installing i series telephones The Nortel Networks i series telephones 120X X can be configured to the network by the end user or by the administrator If the end user is configuring the telephone the administrator must provide the user with the required parameters A maximum of 90 IP telephones including Nortel Networks 12050 Software Phones and H 323 devices such as NetVision handsets can be connected on the Business Communications Manager system if the system resources support the deployment Before installing Before installing the 1200X telephone ensure that e the telephone has the appropriate power supply for your region e if powered locally the installation site has a nearby power outlet otherwise it can be powered through a Power Inline Patch Panel PiPP e the installation site has a 10 100 BaseT Ethernet connection Caution Do not plug the telephone into an ISDN connection This can
21. but additional base units can be added Figure 51 Multiple Business Communications Manager systems network diagram Head Office Warehouse Business Business System System Communications Communications telephone telephon Manager Manager PSTN fallback route Intranet VolP trunk i2050 Software Phone i2004 telephone remote 7 j2004 To set up a network of Business Communications Managers 1 Ensure that the existing network can support the additional VoIP traffic Coordinate a Private dialing plan between all the systems On each Business Communications Manager system e Set up outgoing call configuration for the VoIP gateway e Setup aremote gateway for the other Business Communications Managers or NetMeeting users e Set telephones to receive incoming calls through target lines e Configure the PSTN fallback and enable QoS on both systems 4 Reboot each system IP Telephony Configuration Guide 150 Chapter8 Typical network applications using MCDN This system uses fallback to PSTN so calls can be routed across the PSTN connection if VoIP traffic between the Business Communications Manager systems becomes too heavy If only one of the Business Communication Managers in a network has a line to the PSTN network all public calls from other systems are funneled through the system with the PSTN connection and all communication between the systems occurs over VoIP trunks
22. isssss een RARE RE hn 35 Setting the Global IP published IP issue mr RS 35 Determining the published IP address 00 000 cee 36 Media gateway parameters for IP service 0000 cee eee eee eee 37 b mii eT Tp ccc 39 I telephone JSCOIUS aazaqesuy eeu ed tb HER A d Rar ep eR E Ee edens 40 IP Telephony Configuration Guide 4 Contents Chapter 3 installing IP telephones iuiunascews en ER ERR ARE PR ARE ad aA 41 IP telephony on the Business Communications Manager 42 Configuring Nortel Networks i series telephones lllessssn 42 Preparing your system for IP telephone registration 000 43 Setting IF terminal general settings iius issued ee en c no 43 tiso yek DRE aeo Gand Achaea HUS EE Or paCOHRER PEE D bd e E 45 Choosing a Jitter BUMS ieee we eid dR ace A ERR ERU ERR e HERES 46 Installing i series telephones iusso cusam ku xm RR ee OE ee 46 Before install 25502086 24846 eeetosrntermtercmnanrs mee amma Samy 46 Using s Sport SWIC vosuneeXuocex ira r9erRra Ateri Rr peed needy 46 Connecting the 200K telephones cca ideanvees beets E Ra deena RR 47 Configuring the i20XX telephones to the system 00 eee eee 47 Registering the telephone to the system 000 e cece eee eee 48 Configuring telephone settings llle 48 Troubleshooting IP telephones 00 eee eee eee es 51 Operation WS s pa dca ERE Lal ae qi bl Und deo GRE
23. with call center 151 multiple Business Communications Manager 149 non linear processing 159 other internet resource considerations 162 PSTN fallback 112 remote IP telephone site 152 signaling method 91 transmission characteristics 154 using a gatekeeper 133 Virtual Private Network ID 144 VoIP destination digits 105 107 WAN engineering 158 networks VLAN ports 40 NEW SET 51 no connection IP telephones 52 no speech path IP telephones 52 non linear processing 159 Nortel NVPA changing handset name 82 username 78 NPI TON 94 number of calls usable link bandwidth 155 O OLI VoIP name display 144 one way delay 163 one way speech path IP telephones 52 outbound traffic filter creating 179 Outgoing call configuration 102 107 102 107 overflow setting 118 outgoing calls P Packet delay 201 packet errors reducing 164 loss 154 163 201 queuing delay 163 Packet InterNet Groper see Ping 173 password H 323 terminals list 81 hot desking 999 61 payload size 154 155 156 177 peak bandwidth 155 156 peak traffic 154 157 physical link capacity 154 Ping 173 193 195 planning modules 162 port settings 129 132 ports firewalls 129 legacy networks 132 PPP B W 154 155 156 preferred codec 90 pre installation requirements 46 prerequisites 31 IP telephones 40 keycodes 34 MI IPT MCDN 148 network assessment 33 network devices 32 network diagram 31 resource assessment 33 system configuration 34 P
24. 0 1 and prior FAX over IP is not supported VoIP trunks Remote gateway No support for configurable Transmit and Receive thresholds VoIP routing No support for special call tyoes for MCDN trunks Local National Special No support for Norstar IP trunking No support for BST doorphone T7316E works configures as a T7316 WARNING This telephone reconfigures when the system is upgraded to BCM 3 5 or newer software No support for key interface module KIM 3 5 and prior Long tones do not work over IP trunks Firewall Default Rules when enabled block call processing and signaling You must add an additional rule to pass Protocol TCP UDP Destination Port H 323 for speech path to initialize If an IP Telephony Remote Gateway IP address is pointed at a Wan Link Interface which is a Published IP address the ISDN WAN Backup Feature will not support VoIP Traffic from any set type to that Published IP Address in some Network Topologies Gatekeepers No support for NetCentrex backup gatekeepers adjustable Call Signaling or RAS ports enabling disabling H245 tunneling Media parameters H323 trunks No support for payload size adjustments Media Gateway parameters No support for T 38 UDP redundancy No support for private received numbering over MCDN P0609327 02 Interoperability 187 Table 47 Software interoperability restrictions and limitations for IP trunking Continued Software release Des
25. 2 1 0 1 gatekeeper limitations ECS does not support fast start in the Call Setup Q 931 and Call Control H 245 routing mode P0609327 02 Interoperability 185 Table 47 Software interoperability restrictions and limitations for IP trunking Continued Software release Description of restriction limitation 3 0 3 0 1 GA Gatekeeper e Officially Business Communications Manager supports RadVision ECS 2 1 0 1 and CSE 1000 as gatekeepers It does not support the Radvision Dialing plan package e Radvision ECS 2 1 0 1 gatekeeper limitations ECS does not support fast start in the Call Setup Q 931 and Call Control H 245 routing mode Note M1 IPT required for networks with Business Communications Managers running 3 5 or newer software does not support a Radvision gatekeeper Call signaling By selecting GatekeeperRouted or GatekeeperResolved you switch Business Communications Manager to gatekeeper mode which means your Remote Gateway table will no longer be a part of your call routing plan Choosing one of the modes will advertise a Business Communications Manager preference The Gatekeeper is the final decisionmaker It will select the mode routed or resolved based on its configuration e GatekeeperRouted routes the Call Setup Channel and Control Channel through the ECS In ECS terminology this mode is called Call Setup Q 931 and Call Control h 245 e GatekeeperResolved routes the Call Setup Channel and
26. 22 65 sched oes Foe eu STROM ARR TR eR ORE RON ew de de urs 191 Measuring intranet OOS aka aes ey acce ORC ORO e ECC e RL RR REC OD clon 193 Measuring end to end network delay lslseele eese 193 Measuring end to end packetloss 0 c eee ete eee 194 Ip HB OUE LL cack seat taal dorada dt drap palin A ead dad ox a anh eee 194 Adjusting Ping measutemenis 2 240h0084 45004440003 RE TEE PRES OER 195 Adjustment Tor processing 44494 454 ERR RN RETRO CEA eee eed e CR CR CR A 195 E o q p ocio nC PTT 195 Measurement procedure 0 0 cee nn 196 Other measurement considerations 0 0 c eee ee ee 196 Decision does the intranet meet IP telephony QoS needs 197 Implementing QoS In IP networks isocisenkbdostkbeoiRiora rk bs Vaawd ves 198 WAM S CT PI EEUU 198 P0609327 02 Contents 9 TOF AID DENA Less Save d QR oss IHR eV e Ie b Prep ecd bd up b 199 Business Communications Manager router QoS support 199 Network Quality of SerViGB oo onc coenae cukk ouem cx do dcm CR DE Rom on KR X dk 200 Neier monter acacdcecelavsriagte ieee Roce esta pde dE d dd 200 Quality of Service parameters 2 cs ese deeb eee seeds Rhet 201 goleo doaa Msg TT 201 S1 ressar serseri arrar raa aea eaa 203 o Je COUTE 211 IP Telephony Configuration Guide 10 Contents P0609327 02 11 Figures Figure 1 NOSHOIK dagan sere pra YO yeas thane RERO EYE HR ER ERROR 24 Pee 2
27. 723 gt G 729 gt G 711 the more resources are set aside for Media Gateways This may result in calls failing to go through because of lack of available resources T 38 UDP Redundancy Default 0 transmitted only once This setting defines the number of times the system will transmit a UDP packet over the network This setting acts as an error control mechanism for unreliable networks by providing the same information more than once with the intention that at least one of the copies will transmit correctly WARNING Each redundancy requires the same amount of bandwidth as the original message This means that a redundancy of 3 requires four times the bandwidth of a single transmission For a more detailed descriptions of the media gateway or other information about the media services card MSC settings for the Business Communications Manager refer to the Programming Operations Guide MSC section P0609327 02 Chapter 2 Prerequisites checklist 39 VoIP trunks Complete this section if you are configuring VoIP trunks Table 9 VoIP trunk provisioning Prerequisites Yes No 7 a Have you confirmed the remote gateway or Gatekeeper settings and access codes required H 323 and SIP trunks 7 6 Have you determined the preferred codecs and payload sizes required for each type of trunk and destination 7 c Have you determined how you are going to split your VoIP resources betwe
28. 729 Highest supported by both ends up to 30 ms Speech path setup methods Business Communications Manager version 3 5 and newer software initiate calls using H 323 fastStart methods The Business Communications Manager will accept and set up calls that have been initiated by another endpoint using H 323v2 fastStart methods as well as H 323 slowStart methods P0609327 02 Interoperability 177 Media path redirection Media path redirection occurs after a call has been established when an attempt is made to transfer to or conference in another telephone To ensure that call transfers and conference works correctly the following rules must be followed e The first preferred codec for VoIP Trunks must be the same on all Business Communications Managers See Configuring media parameters on page 89 If this codec is 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 1 IPT the profile on the IPT 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 IPT payload size must be set to 30 ms If these rules are not adhered to simple calls will still go through but some transfer scenarios will fail Gatekeeper The Business Communicati
29. Common Settings Schedule Names Refer to the Programming Operations Guide for detailed instructions about renaming schedules if required 2 Open Services Telephony Services Scheduled Services Routing Service and click VoIP The VoIP schedule screen appears in the right frame VoIP Service setting Manual v Overflow Y Y 3 Change the Service setting to Manual P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 119 4 Change the Overflow setting to Y Activating the VoIP schedule for fallback Before activating the VoIP schedule calls using the destination code are routed over the PSTN This is because the system is set to use the Normal schedule which routes the call over the PSTN Once the VoIP schedule is activated calls made with the VoIP destination code are routed over the VoIP trunk The VoIP line must be activated FEATURE 873 from the control set for the VoIP trunk which is specified when the trunk is created Services Telephony Services Lines VoIP lines Enabled VoIP lines Line XXX General For information about control sets and configuring VoIP line records refer to the Programming Operations Guide 1 Dial FEATURE 873 from the control set for the VoIP trunk The phone prompts you for a password Type the password Press OK The first schedule appears Scroll down the list until VoIP is selected 5 Press OK The VoIP schedule stays active even after a syst
30. Control Channel directly to the far end without ECS intervention In ECS terminology this mode is called Direct By using this method you will speed up you call setup time This is the recommended configuration for the Business Communications Manager ECS Configuration e Accept calls this must be enabled so that calls pass through the ECS Gatekeeper e Routing Mode it is recommended that you set this to Direct to minimize call setup time The Business Communications Manager also supports routing of Setup Q 931 and Call Control H 245 Important The Business Communications Manager does NOT support the second option the routing of Setup Q 931 The option Check that call is active every XXX seconds must be unchecked e Force Direct For Service Calls this setting on the Settings Advanced tab should be enabled if the ECS Gatekeeper has been configured to use Direct call routing ITG version 26 26 does not include support for gatekeeper interaction To be able to establish calls between Business Communications Manager 3 0 and ITG through a gatekeeper follow the configuration steps found in the Using a gatekeeper on page 133 3 0 1 and prior If these systems are running in a private network with systems running BCM 3 5 or newer software they must have QoS patch 3 0 0 25 or later installed to allow H 323 VoIP trunking to function correctly 3 0 1 and prior SIP trunks SIP trunks can only be set up between tw
31. Endpoint Properties dialog displays C Ensure the following fields are set Table 33 Radvision Predefined Endpoints Properties settings Field Value Description Endpoint Type Gateway Force Online Status check box selected Registration IP ip address This is the IP address of the Meridian IPT system Aliases Add Name The name of the IPT that will be Name displayed Phone Number Phone Number The number assigned to the IPT Radvision uses this number to identify calls to be routed to this IPT Allowed Services Allowed Ensure the IPT service is on the list and is Disallowed Allowed 8 Close the application 9 Run system tests to ensure the gatekeeper is routing calls correctly IP Telephony Configuration Guide 136 Chapter 7 Optional VoIP trunk configurations Using CSE 1000 as a gatekeeper Both the Business Communications Manager and the CSE 1000 must be set to the parameters described in this section for the gatekeeper to work effectively The CSE 1000 GK Admin tool is obtained from http lt Gatekeeper IP gt gk Before an endpoint registers with the CSE 1000 gatekeeper it must first be added to the gatekeeper configuration Before a registered endpoint may make calls it must have its numbering plan information assigned within the gatekeeper configuration Before any of these configuration changes become part of the gatekeeper active configuration they must be committed to the ac
32. GOD IP SOINS voit eee ditur a e deo pg dun od desde aT ding too ees 35 Figure 3 Selecting the Published IP address seseeseen e 36 Figure4 System Configuration Parameters screen 0 00 cee eee 37 Figure5 Set registration properties 0 0c eee 43 Figure 6 IP terminal registration server status 2 00 c eee eee 56 or TAMEN Terminal SIGS rp ee ee de ee Riad aces 57 Figures IP Terminal status dialog DOX 6 2465 denne ees i Rar AR RR ER ERE DROPS Rp 58 Figure9 IP Telephony Features List 0 0 00 e eee ees 59 Figure 10 Add Modify Telephony Features List 0 00 eee eee ee ees 60 Figure 11 IP Terminal Status tab list 2 00 ccccc cee een eed anes mme ens eed eee s 61 Figure 12 Labelset delauliS cc ccs aceite vada deh Peed T RMS e REG SO RN OR AES 64 Figure 13 Deregister DN from Configuration menu 0 00 eee eee eee 67 Figure 14 Deregister DN from Configuration menu 0 0 eee 68 Figure 15 12050 Communications Server isresirieinsierini ke ARR ACABA x 71 Figure 15 OSO SIG Wie 2sesz op PSTSELOSRAGGq Xa xc Rd d d adde x dar d de 72 Figure 17 Defining Codec and Jitter Buffer for all terminals 00 76 Figure 18 Viewing the Summary tab for H 323 terminals l l 0000 eae 77 Figure 19 H 323 Terminal list dialog DOX cssc 9 Rm Re RR Rack ela 79 Figure 20 H 323 Terminal list with terminal information llle 81 Figure 21 Deregister DN
33. Hot desking feature All your telephone features and setup can travel with you between offices Simplicity and consistency A common approach to service deployment allows further cost savings from the use of common management tools resource directories flow through provisioning and a consistent approach to network security As well customers can centrally manage a host of multimedia services and business building applications via a Web based browser The ability to network existing PBXs using IP can bring new benefits to your business For example the ability to consolidate voice mail onto a single system or to fewer systems makes it easier for voice mail users to network e Compatibility Internet telephony is supported over a wide variety of transport technologies A user can gain access to just about any business system through an analog line Digital Subscriber Line DSL a LAN frame relay asynchronous transfer mode SONET or wireless connection IP Telephony Configuration Guide 22 Chapter1 Introduction e Scalability A future proof flexible and safe solution combined with high reliability allows your company to focus on customer needs not network problems Nortel Networks internet telephony solutions offer hybrid environments that leverage existing investments in Meridian and Norstar systems e Increased customer satisfaction Breakthrough e business applications help deliver the top flight customer service that lea
34. If the frames arrive at the other end at the same rate voice quality is perceived as good In many cases however some frames can arrive slightly faster or slower than the other frames This is called jitter and degrades the perceived voice quality To minimize this problem configure the IP telephone with a jitter buffer for arriving frames Note You can only change the jitter buffer on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone This is how the jitter buffer works Assume a jitter buffer setting of five frames The IP telephone firmware places the first five arriving frames in the jitter buffer e When frame six arrives the IP telephone firmware places it in the buffer and sends frame one to the handset speaker e When frame seven arrives the IP telephone buffers it and sends frame two to the handset speaker The net effect of using a jitter buffer is that the arriving packets are delayed slightly in order to ensure a constant rate of arriving frames at the handset speaker This delaying of packets can provide somewhat of a communications challenge as speech is delayed by the number of frames in the buffer For one sided conversations there are no issues However for two sided conversations where one party tries to interrupt the other speaking party it can be annoying In this second situation by the time the voice of the interrup
35. Manager IP Telephony supports H 323 hop count This is the number of routers the signal must go through to reach the destination device The more hops that are required the more potential there is for voice quality issues to arise hot desking See Feature 999 hub Center of a star topology network or cabling system IEEES02 ESS This is the LAN and switch standard used to define the connection between the access point and the NetVision handset onto the network The handset is given the ID code of the device s with this standard so the access points recognize them 12050 Software Phone This is a computer based version of an IP telephone Once installed it acts and is programmed as you would the 12004 telephone You must have a sound card and a USB headset to use this application interoperability Interoperability refers to how compatible Business Communications Manager data configuration is with the rest of the network Business Communications Manager IP Telephony adheres to the ITU T H 323v2 standards and is compatible with any H 323v1 or H 323v2 endpoints This also refers to IP compatibility issues between released versions of the Business Communications Manager Business Communications Managers on the network with earlier versions of the software will not have the same operability for VoIP trunks as systems with 3 5 software IP server On the Business Communications Manager this is the server that registers IP
36. Manager can have up to two LAN connections P0609327 02 Chapter 1 Introduction 27 Public Switched Telephone Network The Public Switched Telephone Network PSTN can play an important role in IP telephony communications In many installations the PSTN forms a fallback route If a call across a VoIP trunk does not have adequate voice quality the call can be routed across PSTN lines instead either on public lines or on a dedicated ISDN connection between the two systems private network The Business Communications Manager also serves as a gateway to the PSTN for all voice traffic on the system Key IP telephony concepts In traditional telephony the voice path between two telephones is circuit switched This means that the analog or digital connection between the two telephones is dedicated to the call The voice quality is usually excellent since there is no other signal to interfere In IP telephony each IP telephone encodes the speech at the handset microphone into small data packets called frames The system sends the frames across the IP network to the other telephone where the frames are decoded and played at the handset receiver If some of the frames get lost while in transit or are delayed too long the receiving telephone experiences poor voice quality On a properly configured network voice quality should be consistent for all IP calls The information under the following headings describe some of the components tha
37. Manager to identify a handset If you need to change the name of an assigned handset 1 Delete the existing record Refer to Deleting a NetVision telephone from the system on page 82 Enter a new record with the new name Adding a NetVision record in the Unified Manager on page 79 You can assign the existing DN to the new record To maintain security assign a new password Changing the DN record of a handset If you need to change the DN number for a handset use the Unified Manager Services Telephony Services General Change DN The change will automatically be reflected in the H 323 Terminals record for the handset When you use the Change DN feature the DN settings are transferred to the new DN and the system features remain active on the new DN Warning Deleting an H 323 Terminals record removes the DN from the Active DNs list This means that system features such as Call Forward No Answer also become inactive Deleting a NetVision telephone from the system If you want to stop a terminal from having access to the Business Communications Manager you can delete the DN record for the terminal 1 2 3 4 5 In the Unified Manager click the keys beside Services IP Telephony and IP Terminals Click H 323 Terminals Click the H 323 Terminal list tab then click on the terminal record you want to delete Click on Configuration and choose Delete Entry A message appears that asks you to confir
38. Set IP 49 signaling method 91 silence compression 176 about 167 comfort noise 172 full duplex 170 half duplex 168 silence suppression full duplex links 157 half duplex links 156 SIP fallback setting 93 site pairs 194 SL 1 MI IPT 25 MCDN fallback 148 MCDN over VoIP 105 SLI Gateway Protocol 95 slow connection IP telephones 52 Sniffer 173 source gateway 172 specifications H 323 gateway 176 speech packets silence compression 167 speech path setup 176 SRG MCDN Zone ID 144 Virtual Private Network ID 144 static IP list NetVision 75 status H 323 terminals list 81 Succession MCDN Zone ID 144 Virtual Private Network ID 144 SWCA group answering 98 switches network prerequisites 31 Symbol see NetVision 175 Symbols 15 system configuration Business Communications Manager prerequisites 34 System wide Call Appearance see SWCA 98 T T 38 fax 142 target lines VoIP trunks incoming calls 98 TCP traffic behavior 199 template file H 323 terminals list 81 terminal status 57 text conventions 16 time exceeded 173 TimeToLive 94 tips 15 Traceroute 173 194 IP Telephony Configuration Guide 220 Index traffic network loading 160 network mix 198 WAN link resources 153 transfer media path redirection 177 transmission characteristics 154 transmit fallback threshold 192 transmit path 156 Transmit Threshold 103 124 troubleshooting IP telephones 51 network delay and error an
39. Setthe Absorbed length to absorb the amount of the destination code that is not part of the DN for the other system If this is a private network PSTN line and the network uses a CDP dialing plan and the remote system identifier is 2 which is part of the remote system DN structure and you specified destination digit of 2 for the remote gateway then configured a destination code of 82 set this field to 1 so that the 2 is still part of the dialout If the destination code is different from the private access code destination digits for the remote system UDP dialing plan or this is a public PSTN enter private access code or the public access number to the remote system into the External field on the route record In this case set the absorbed length to the number of digits in the destination code The system will dial out the External you entered in front of the rest of the number that the user dialed Setting up the VoIP schedule to overflow Once you have configured the routing and destination codes ensure that the Routing Service schedule allows fallback Overflow and allows you to activate the service from a control set You will note that the Routing Service does not have a Normal schedule This is because the Normal schedule is the schedule that runs when no routing services are active Follow these steps to set up the VoIP schedule for routing services 1 Rename Schedule 4 to VoIP Services Telephony Services Scheduled Services
40. Software Phone 1 0 x Nortel Networks i2001 Nortel Networks M1 IPT 3 0 or 3 1 Microsoft NetMeeting 3 0 Symbol NetVision Telephone 03 50 12 01 00 24 or greater Nortel Networks Norstar IP Gateway Nortel Networks Succession with CSE1k 3 0 IP Telephony Configuration Guide 176 Interoperability Business Communications Manager IP Telephony interoperates with the Gatekeeper applications Radvision ECS 3 2 CSE 1000 and NetCentrex which conform to the specifications in the following tables Table 43 Engineering specifications Capacity 1 to 8 ports Voice compression G 723 1 MP MLQ 6 3 kbit s or ACELP 5 3 kbit s G 729 CS ACELP 8 kbit s supports plain Annex A and Annex B G 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 signaling DTMF TIA 464B Call progress Speech path setup methods Call Initiator e H 323 fastStart Call Terminator e H 323 slowStart e H 323v2 fastStart End to end DTMF signaling digits 0 9 and fixed duration tones only Meridian 1 IPT does not support the Radvision gatekeeper Table 44 Supported voice payload sizes 30 ms in 10 ms increments Codec Receive transmit to M1 IPT Receive transmit to others G 711 Highest supported by both ends up to 30 ms 30 ms in 10 ms increments G 723 1 30 ms 30 ms G
41. Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active G 729 A codec that provides near toll quality at a low delay Uses compression to 8 kbit s 8 1 compression rate 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 Codecs with VAD Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active G 723 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 The G 723 CODEC should be used only with third party devices that do not support G 729 or G 711 Codecs with VAD Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active IP Telephony Configuration Guide 206 Glossary gatekeeper A gatekeeper is server application on a network that tracks IP addresses of specified devices to provide routing and optionally authorization for making and accepting calls for those devices The Business Communications Manager supports RadVision CSE 1000 and NetCentrex gatekeeper applications H 323 The ITU standard for multimedia communications over an IP network Business Communications
42. WAR RL dee ALN A 52 Configuring DHCP i usucsiancuum xd x RR x wPex e ex d Ee Ree eRe dd 53 Setting up DHCP to work with IP terminals 00 000 e eee 53 IP telephony DHCP NOS 6245085605445 RE GO XN RR KR RRGXAGUERR EH ORES stut 54 Checking IP Server Status uiuis eem xen dh Ix 3e RR UE Oye Ree o dx 56 Modifying IP telephone status settings lille 57 Working with the features NS PPP D ia 59 Using the Services button to access features 00000 illl 60 Resetting the Hot Desking password 0 00 cee ees 61 Notes about Hot Desking iussu m9 m9 Rhe ko ean 62 Customizing feature labels 000 eee 63 Changing features or labels on the memory buttons 2 22255 64 Download firmware to a Nortel IP telephone 0 000 eee 65 Forcing a firmware download to an IP telephone llllleelsssss 65 Deregistering DNs for IP telephones 000000 cece eee eee 66 Deregistering a telephone using the IP record 2 20000eeeeee 66 Deregistering a telephone using a DN registration heading 67 Moving IP TelephofteS uuues sce cm dk oon qme ener qe Regex e c Rte kw n 69 Moving IP telephones and retaining the DN 0 2020000 ee aee 69 Moving telephones and changing the DN llseslllee eese 69 Configuring a new time zone on a remote telephone llus 70 Nortel Networks i2050 Software Phone 000 e e
43. all packets on the data network IP Telephony Configuration Guide 174 Network performance utilities P0609327 02 175 Appendix D Interoperability This section discusses interoperability between the Business Communications Manager and other networks including e Speech path setup methods on page 176 e Media path redirection on page 177 e Gatekeeper on page 177 e Asymmetrical media channel negotiation Net Meeting on page 178 e Setting up Remote Routers for IP Telephony Prioritization on page 179 e Using VLAN on the network on page 181 e Symbol NetVision telephones on page 183 e Software interoperability compatibility and constraints on page 183 Business Communications Manager 3 6 IP Telephony adheres to the ITU T H 323v2 standards Such endpoints include the Nortel Networks M1 IPT and Microsoft NetMeeting As well the Business Communications Manager is backward compatible and interoperates with the Nortel Networks i series 200X telephones and 12050 Software Phone and with the Symbol NetVision IP phones The following table summarizes this information Table 42 Business Communications Manager 3 6 IP Interoperability Summary Vendor Product Version Nortel Networks Business Communications Manager 2 5 FP1 MR1 or greater with QoS patch 3 0 0 25 or greater Nortel Networks i2002 i2004 3002B20 or greater Nortel Networks i2050
44. an identifying domain name for your SIP trunks Transport UDP read only This setting refers to the way the Business Communications Manager internally processes the trunk packets Do not confuse this setting with the UDP dialing rule IP Telephony Configuration Guide 96 Chapter5 Configuring local VoIP trunks Notes about NPI TON aliases for H 323 trunks NPI TON aliases store dialed number prefixes as well as information about the type of number A dialed number can be qualified according to its TON type of Number as well as its NPI numbering plan identification Nortel Networks recommends this format over the E 164 format for encoding dialed numbers and aliases registered with a gatekeeper When using a gatekeeper and attempting to place an outgoing VoIP trunk call ensure that the route and dialing plan configuration matches the NPI TON aliases registered by the destination with the gatekeeper These requirements are summarized in the following table Table 24 Route and Dialing Plan configurations for NPI TON Route DN type Dialing Plan used by calling gateway Alias configured for calling gateway Public Public PUB lt dialedDigitsPrefix gt Private Private Type None PRI lt dialedDigitsPrefix gt Private Type CDP CDP lt dialedDigitsPrefix gt Private Type UDP UDP lt dialedDigitsPrefix gt P0609327 02 Chapter 5 Configuring local VoIP trunks
45. available to the caller Pre installation system requirements Ensure that you have obtained the following information or familiarize yourself with the requirements before continuing with VoIP trunk configuration Keycodes Before you can use VoIP you must obtain and install the necessary keycodes See the Keycode Installation Guide for more information about installing the keycodes Talk to your Business Communications Manager sales agent if you need to purchase VoIP keycodes Each keycode adds a specific number of VoIP trunks You must reboot your Business Communications Manager after you enter VoIP keycodes to activate trunking You then must identify each trunk as either H 323 or SIP trunks Refer to Counting IP trunks on page 87 If you want to use the MCDN features on the VoIP trunks you will need an MCDN keycode If you have already deployed MCDN for your SL 1 PRI lines you do not require an additional keycode Note SIP trunks do not support MCDN Published IP address You will require the public IP address to set up the gateways for VoIP trunks Refer to Finding the published IP address on page 35 for details SIP network data considerations If you plan to use SIP trunking ensure that your IP network is set up to accommodate the restrictions and requirements Refer to the NAT Firewall and QoS sections of the Programming Operations Guide for data programming details for these utilities To view a general list o
46. bandwidth consumption by voice data packets Note If the IP telephones are using VoIP trunks for the call the codec set for the trunks overrides the telephone settings For IP telephones the Business Communications Manager supports both a law and mu law variants of the G 711 CODEC as well as the G 729 and G 723 CODECS e 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 Choose the G 711 CODEC with the companding law alaw or ulaw that matches your system requirements e The G 729 CODEC samples the voice stream at 8Kbps The voice quality is slightly lower using a G 729 but it reduces network traffic by approximately 80 e The G 723 CODEC should be used only with third party devices that do not support G 729 or G 711 e Codecs with VAD Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active Note You can only change the codec on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone IP Telephony Configuration Guide 46 Chapter3_ Installing IP telephones 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
47. be able to communicate throughout the Bridged LAN Choosing DHCP for VLAN If you use a DHCP server remote to your Business Communications Manager you must enter any VLAN IDs manually on IP telephones By using the Business Communications Manager DHCP server you can configure DHCP to auto assign a VLAN ID to each IP telephone that registers With this configuration you can also choose to manually enter VLAN IDs if you choose The Business Communications Manager DHCP server becomes the default VLAN that everyone can reach The server provides the network configuration information in the default VLAN and it also provides the VLAN information for the network Refer to the Business Communications Manager Programming Operations Guide for the DHCP settings for VLAN Refer to Configuring the 120XX telephones to the system on page 47 for information about configuring VLAN on the Nortel IP telephone IP Telephony Configuration Guide 182 Interoperability Assigning VLANs becomes important if you have multiple devices connected to the same switch port such as when you use a 3 port switch to connect a computer and IP phone on the same network cable In this case the system needs to apply the correct VLAN for each device Specifying the site specific options for VLAN The Business Communications Manager DHCP server resides in default VLAN and is configured to supply the VLAN information to the IP phones The DHCP server will supply site spe
48. by the administrator using spreadsheets and other statistics packages The installer can check if the intranet network management software has any delay measurement modules which can cause a delay distribution measurement for specific site pairs IP Telephony Configuration Guide 194 Quality of Service Delay characteristics vary depending on the site pair and the time of day The evaluation of the intranet includes taking delay measurements for each site pair If there are important changes of traffic in the intranet include some Ping samples during the peak hour For a more complete evaluation of the 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 13 ms 1 m
49. carrying capacity of the 56 kbit s link can be 28 kbit s and for the T1 1 3056 Mbit s In some IP Telephony Configuration Guide 154 Efficient Networking organizations the thresholds can be lower than those used in this example In the event of link failures spare capacity for rerouting traffic is required Some WAN links can exist on top of layer 2 services such as Frame Relay and Asynchronous Transfer Mode ATM The router to router link is a virtual circuit which is subject not only to a physical capacity limits but also to a logical capacity limit The installer or administrator needs to obtain the physical link capacity and the QoS parameters The important QoS parameters are CIR committed information rate for Frame Relay and MCR maximum cell rate for Asynchronous Transfer Mode ATM The difference between the current capacity and the acceptable limit is the available capacity For example a T1 link used at 48 during the peak hour with a planning limit of 85 has an available capacity of approximately 568 kbit s Network engineering This section describes some network engineering criteria that you need to consider for your system Refer to the information under the headings below for details e Bandwidth requirements on half duplex links on page 155 Bandwidth requirements on full duplex links on page 156 e LAN engineering examples on page 157 e WAN engineering on page 158 e QoS Monitori
50. cause severe damage to the telephone Plug the telephone only into a 10 100 BaseT Ethernet connection e if you are using an IP telephone that does not have a 3 port switch ensure you have 10 100 BaseT Ethernet connections for both the telephone and for your computer equipment Using a 3 port switch In an office environment where a LAN network already exists most computers will already be connected to a LAN line To avoid the necessity of installing duplicate network connections you can use a Nortel Networks 3 port switch for older model 12004 telephones This switch allows the telephone and computer to connect to the same network connection For more information consult the 12004 setup card and the 3 way switch documentation P0609327 02 Chapter 3 Installing IP telephones 47 The 12001 12002 and newer models of the 12004 telephone have an adapter in the telephone housing that replaces the requirement for this switch Connecting the i200X telephones Follow these steps to connect a Nortel IP 1200X telephone 1 2 3 Connect one end of the handset cord to the handset jack on the telephone base Connect the other end of the handset cord to the handset Connect one end of a Cat 5 line cord with RJ45 connectors to the line cord jack on the telephone base Connect the other end of the line cord to the Ethernet connection or to the 3 way switch connector Note Newer 1200X terminals have a 3 way switch built into the t
51. click the Services IP Telephony IP Trunks keys 2 Click SIP Trunks The Summary dialog appears Figure 28 SIP Summary dialog box 9 Q IP Telephony SAY Q System Configuration IP Terminals Name siP UA 9 Q IP Trunks 9 Q H 323 Trunks Stamm Remote Gateway 0 0 9 Address Book Description VoIP SIP Gateway Incoming calls Assigning target lines To receive an incoming call directly to the telephone from a VoIP network you need to ensure that the telephone is mapped to a target line How to use target lines 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 e You can assign the target line to a number of telephones if you want the call to be answerable to a call group for instance e If System Wide Call Appearance SWCA keys are configured on memory buttons on the telephones the incoming line acts the same way as any other incoming call which depends on how SWCA has been set up to behave Refer to the Programming Operations Guide and the Telephony Feature Handbook for more information about setting up SWCA keys e You can assign the target line number to a Hunt Group DN if you want the call to appear on a group of telephones set up as a hunt group Refer to the Programming Operations Guide for more information about setting up Hunt groups P0609327 02 Chapter 5 Configuri
52. codes 116 MCDN 147 MCDN networking 148 Mean Opinion Score 201 MOS for codecs 201 scheduling 118 using PRI line 123 Fallback to Circuit Switched Local Gateway 93 fastStart 178 FAX over IP 142 FEATURE hot desking 999 62 features list 59 services key 900 60 filtering criteria 180 ranges 180 firewall IP configuration note 52 firewalls configuring 129 network prerequisites 32 ports 129 firmware downloading to IP telephones 65 Force Direct for Service Calls Radivision 134 force download 65 Force Online Status Radvision 135 FR B W 154 155 156 Frame Relay 154 full duplex link bandwidth requirements 156 silence compression examples 170 silence suppression 157 VoIP load 161 WAN engineering 158 G G 711 154 155 156 Q 723 1 154 155 156 G 729 154 155 156 Gatekeeper interoperability support 135 Radivision ECS 2 1 0 1 134 gatekeeper 133 call scenarios 141 defined 25 interoperability 177 network prerequisites 31 signaling method 91 Gatekeeper IP Local Gateway 94 GateKeeperResolved 93 GateKeeperRouted 93 gateway Business Communications Manager QoS support 199 connecting to intranet 162 destination digits 117 118 H 323 specifications 176 IP telephones 49 monitoring QoS 192 network prerequisites 31 progress tones 178 remote configuring 103 106 Gateway Protocol 103 Gateway Protocol Local Gateway 95 Gateway Type 103 Global IP see Published IP address 35 GWProtocol 95
53. commit them to the active database from the Gatekeeper Admin tool as described below 1 Select GK Standby DB Admin 2 Select Database Actions 3 Select Single Step Commit and Crossover P0609327 02 Chapter 7 Optional VoIP trunk configurations 139 Configuring Codec Compatibility The default codec settings for a CSE1000 are not compatible with those used by a Business Communications Manager system In order to successfully make IP trunk calls between a Business Communications Manager and the CSE 1000 the codec configuration on both the Business Communications Manager and the CSE 1000 must coincide as shown in the table below As well any configured codecs on the CSE 1000 must have their payload size set to 30 ms Caution The CSE 1000 can only register five codecs at once This can include G 711 mu law G 711 a law T 38 G 711CC and either G 729A G729AB or G 723 1 It is important to that you disable the unused codecs This ensures that the required codecs get registered with the DSP Failure to disable unused codecs could result in the wrong codecs being registered with the DSP which would create call failures Table 36 CSE1000 codec compatibility with endpoints Business Communications Manager preferred codec Refer to Configuring media parameters on page 89 CSE 1000 codec configuration G 729 G 729 AB is enabled silence suppression is enabled G 729A and G 723 are disabled G 729 G
54. data communication network data communications Processes and equipment used to transport signals from a data processing device at one location to a data processing device at another location default gateway For IP telephony this refers to the router that closest to the IP telephone DS30 split This term refers to the allocation of media resources by the media services card MSC on the Business Communications Manager The default setting is a 2 6 split meaning that DS 01 and DS 08 are automatically used internal media processing including IP telephony If you plan to have a maximum number of IP telephones you may need to set your system so that it uses DS30 bus 07 DS30 3 5 split as a processor for internal media traffic including IP telephony instead of for digital traffic through a media bay module enbloc AII dialed digits sent in a single expression The system waits for all digits to be dialed before processing the call ESSID This is the code that identifies the access point that a NetVision handset uses to connect to the internet and the Business Communications Manager fallback to PSTN Your VoIP trunks can be configured to revert to land lines processed over the PSTN public switched telephony network if the IP network experiences quality issues This process occurs during call setup QoS must be active on the network to use this feature FEATURE 900 This feature code accesses a display menu on Nortel IP
55. delimiters will guarantee parsing For example 001 01 and 1 would all be parsed correctly and interpreted as value 0x01 internal to the IP telephone Note that these fields must be no more than three digits long each ppppp is the port number in ASCII encoded decimal It does not need to be five digits long as the and delimiters will guarantee parsing For example 05001 5001 1 00001 etc would all be parsed correctly and accepted as correct The valid range is 0 65535 stored internally in 12004 as hexadecimal in range 0 0xFFFF Note that this field must be no more than five digits long in all cases the ASCII encoded numbers are treated as decimal values and leading zeros are ignored More specifically a leading zero does not change the interpretation of the value to be OCTAL encoded For example 0021 021 and 21 are all parsed and interpreted as decimal 21 IP Telephony Configuration Guide 56 Chapter 3 Installing IP telephones Checking IP server status You can perform a status check on the Business Communications Manager server that gets used to register IP terminals This screen provides information about the server and whether the telephone properly registered 1 Inthe Unified Manager open Services IP Telephony IP Terminals and click Nortel IP Terminals 9 Q Services Telephony Services Doorphones 9 Q IP Telephony System Configuration 9 IP Terminals H 323 Terminals a I
56. determines which lines the handset can access Note NetVision handsets cannot use trunks that have been configured with the SIP protocol The handset can communicate with any other type of telephone supported by the Business Communications Manager system Click on the following headings to view more information about NetVision connectivity Access points Instructions about installing an 802 11b access point are provided with the access point equipment which is sold and installed separately The access point is set up with a unique identifier ESS ID which is entered into the handset either through a configuration download or manually through the dialpad to allow the handset to access the system through that access point IP Telephony Configuration Guide 74 Chapter 4 Installing NetVision telephones Keycodes Before setting up NetVision telephones ensure that you have enough IP client keycodes enabled to register all the NetVision telephones you require For information about entering keycodes see the Keycode Installation Guide IP clients are distributed on a one to one basis with NetVision and IP telephones so ensure that you take your entire system into consideration Handset and call functions Symbol supplies a handset user guide that describes the features on the NetVision handset and how to use them to perform basic functions The Business Communications Manager NetVision Feature card explains how to use the handset
57. different media types for the receive and transmit channels and immediately hangs up a call taken with asymmetric audio channels In this case the party on the Business Communications Manager switch hears a treatment from the switch normally a reorder tone The party on the NetMeeting client loses connection To solve this problem in NetMeeting under the Tools Options Audio Advanced check Manually configure compression settings and ensure that the media types are in the same order as shown in the Business Communications Manager media parameters table The following table lists the names used by the Business Communications Manager local gateway table and the matching names in NetMeeting Table 45 Name comparison Business Communications Manager media parameters table MS NetMeeting G 723 1 6 3 Kbit s MS G 723 6400 bit s G 723 1 5 3 Kbit s MS G 723 5333 bit s G 711 p law CCITT u law G 711 A law CCITT A law No feedback busy station The Business Communications Manager VoIP gateway provides call progress tones in band to the user If a busy station is contacted through the gateway the gateway plays a busy tone to the user However as NetMeeting does not support fastStart no speech path is opened to the user before the call connects Because of this the user on the NetMeeting station does not hear a busy signal from the gateway P0609327 02 Interoperability 179 Setting up Remote Routers for IP T
58. 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 must determine the following e Is there existing QoS What kind IP telephony traffic must take advantage of established mechanisms if possible e What is the traffic mix If the IP telephony traffic is light compared to data traffic on the intranet then IP QoS can work If IP telephony traffic is heavy data services can be affected if QoS is biased toward IP telephony traffic P0609327 02 Quality of Service 199 TCP traffic behavior Most of corporate intranet traffic is TCP based Different from UDP which has no flow control TCP uses a sliding window flow control mechanism Under this design TCP increases its window size increasing throughput until congestion occurs Congestion results in packet losses and when that oc
59. from Configuration menu lsllee ee 83 Figure 22 IP Trunks Settings SCIGB E uccsucs esee kem m yx e qom kk m ee enews 87 Figure 23 H 323 Media Parameters dialog box liliis 89 Figure 24 SIP Media Parameters dialog box sseeeee IA 89 Figure 25 Local gateway IP interface H 323 Trunks 00 eee eee eee 92 Figure 26 Local gateway IP interface SIP trunks 000 eee eee eee 92 Figure 27 SIP Dialing Sub Domain settings 100 000 0008 eerraew wee kms 97 Figure 28 SIP Summary dialog BOX wc cscs cece aces ede edad es eda deed exo 98 Figure 29 Internal call from Meridian 1 tandems to remote PSTN line 102 Figure 30 Remote gateway dialog box 0c cee tenes 104 Figure 31 Add an entry to the SIP address book 0 0 00 106 Figure 32 Calling into a remote node from a public location 044 110 Figure 33 PSTN fallback Tagram scons cede rere scene eer c cede dn ANa 113 Figure 34 Add route dialog DOK nintendo de aix ee RECAP RE inaki SERRA REGE dH S d 114 Figure 35 Route XXX GCSB ceecaxsxeus me odore uo d dade ER Herde RR RC e EE 115 Figure Je VolP schedule 4uocscceiEad eX PE e dhs AR EAE a dR eS wee oo GRA OUR UR CA 117 Figure 37 Normal schedule routing information liliis 118 Figure 38 Setting up routes and fallback for call to remote system CDP dialing code 120 Figure 39 Setting up routes and fallback for remote external call CDP dialing cod
60. has any Business Communications Managers running BCM 3 5 or newer software SIP trunk interoperability issues The following bullets list the restrictions and requirements for using SIP trunks on a Business Communications Manager SIP trunking uses SIP ALG Application Level Gateway which has the following limitations no support for nested NAT no support for non SIP third party NAT no support for domain names that require NAT or firewall translation the application only uses an IP address in URI Uniform Resources Identifiers format no third party SIP endpoints behind Business Communications Manager NAT are supported in this release multiple media types are supported on the same call but multiple codecs for the same media type are not multicast is not supported no encryption decryption is supported within the body of a SIPs message VPN encryption between Business Communications Managers is supported SIP trunks use the UDP signaling protocol on a fixed port 5060 the Business Communications Manager is a SIP UA client only SIP trunks are not supported across a NAT boundary as they assume the Business Communications Manager published and public IP addresses are the same address SIP call forming is not supported P0609327 02 Interoperability 189 e SIP trunks do not support the MCDN networking protocol e Business Communications Manager call redirection and conferencing are supported e a third party SIP pa
61. header specifies a level of priority service This is quality of service routing For QoS to be successful for IP telephony it must be end to end on the network P0609327 02 Glossary 209 Network note Any systems in a private network that are running software versions previous to BCM 35 or later software must have a QoS patch installed to allow them to be compatible with the H 323 version introduced in the BCM 3 5 software RAS Registration and Admittance Services RAS are a gatekeeper function NetCentrex does not use RAS silence compression silence suppression This is the utility that omits the data packets that occur when no one is talking during the IP trunk calls thus reducing the bandwidth load required for IP calls Symbol NetVision handsets These IP telephones connect to the system through wireless access points connected to the same network to which the Business Communication Manager is connected T 38 fax Refer to VoIP Fax target lines These are internal channels on the Business Communications Manager that allow you to direct incoming calls to specific telephones call groups Hunt groups or system devices System telephones require target lines if they have not already been configured when receiving VoIP trunk calls so the call knows where to go terminal Device capable of sending or receiving data over a data communications channel throughput Indicator of data handling ability Measures dat
62. leave this field blank 4 Inthe DN type box choose Public VoIP line pool 1 On the navigation tree click the route you created for the VoIP lines 2 In the Use Pool field type the letter of the line pool for the VoIP lines 3 Leave the External field blank unless the destination digit you are using for the remote gateway is different than the number you want to use for the destination code 4 Inthe DN type box choose Private Go to the next section Adding the destination code for the fallback route on page 116 Adding the destination code for the fallback route Create a destination code that includes the VoIP and PSTN routes that you created in Adding routes for fallback on page 114 to respond to the same access number destination code When this code is dialed the Business Communications Manager will select the VoIP line if possible If the line is not available the call will fall back to the PSTN line As well you need to create or ensure that your destination code 9 includes a Normal and VoIP schedule that includes the route you created to the local PSTN Note If you already have a line pool access code defined as 9 you will need to delete this record before you create the destination code Follow these steps to create destination codes for your fallback route 1 Open Services Telephony Services Call Routing and highlight Destination Codes 2 Click Add The Add Destination codes dialog box appears Add D
63. one is specified by the gatekeeper Refer to the gatekeeper software documentation for information about changing IP addresses Gatekeeper notes e The Business Communications Manager has been tested by Nortel Networks to be compliant with RADVISION ECS 3 2 GK http www radvision com and CSE 1000 gatekeeper applications e A gatekeeper may help to simplify IP configuration or the Business Communications Manager dialing plan however it will not simplify the network dialing plan IP Telephony Configuration Guide 134 Chapter 7 Optional VoIP trunk configurations M1 IPT does not support a RadVision gatekeeper Keep this in mind if you have an M1 in your private network Using Radvision ECS 3 2 GK as the gatekeeper When you use Radvision ECS 3 2 GK as the gatekeeper with the Business Communications Manager use the configurations described in this section For detailed information about Radvision and how to open and use the application refer to the documentation for the application Configuring Radvision for Business Communications Manager 1 2 3 4 Open the Radvision application On the viaIP Administrator screen select the Settings tab then click on the Basics button Beside the Who can register field choose Everyone In the left frame click the Calls button Ensure the following fields are set Table 31 Radvision Calls screen required settings Field Value Descripti
64. 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 IP Telephony Configuration Guide 164 Efficient Networking Reducing hop count To reduce end to end delay reduce hop count especially on hops that move across WAN links Some of the ways to reduce hop count include e Improve meshing Add links to help improve routing adding a link from router to router4 instead of having the call routed from router1 to router2 to router3 to router4 reducing the hop count by two e Router reduction Join co located gateways 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
65. packets Late packets that are lost are replaced with silence decreasing quality Increase the size of the jitter buffer to improve quality when jitter is high Reduce packet errors Packet errors in intranets correlate to congestion in the network Packet errors are high because the packets are dropped if they arrive faster than the link can transmit Identify which links are the most used to upgrade This removes a source of packet errors on a distinct flow A reduction in hop count provides for less occurrences for routers and links to drop packets Other causes of packet errors not related to delay are as follows e reduced link quality e overloaded CPU saturation e LAN saturation limited size of jitter buffer If the underlying circuit has transmission problems high line error rates outages or other problems the link quality is reduced Other services such as X 25 or frame relay can affect the link Check with your service provider for information Find out what the router threshold CPU utilization level is and check if the router conforms to the threshold If a router is overloaded the router is continuously processing intensive tasks Processing intensive tasks prevents the router from forwarding packets To correct this reconfigure or upgrade the router A router can be overloaded when there are too many high capacity and high traffic links configured on it Ensure that routers are configured to vendor guidelines
66. start 75 changing name for handset 82 common codec 183 configuration process 78 connectivity 73 deleting handset 82 DHCP static IP list 75 DN records 82 H 323 Terminals record 79 installing 73 80 interoperability 175 model 80 name restrictions 78 serial cable 75 supporting documentation 74 unique name 80 updating H 323 record 81 network adjust jitter buffer 164 adjusting Ping measurements 195 analysing QoS needs 197 assessment prerequisites 33 asymmetrical media channel negotiation 178 devices prerequisites 32 DiffServ 199 implementing 162 insufficient link capacity 162 late packets sampling 195 P0609327 02 Index 217 link delay 163 loading 160 locations prerequisites 31 monitoring 200 planning modules 162 port settings 132 post installation measurements 165 quality of service 200 recording routes 194 reducing hop count 164 reducing packet errors 164 Sniffer 173 TCP traffic 199 traffic mix 198 troubleshooting routing 165 voice quality codec for IP telephones 45 networking additional feature configuration 159 Business Communications Manager prerequisites 34 call chain configuration 151 checklist for IP telephony 31 delay and error analysis 162 determining bandwidth 153 determining WAN link resources 153 efficiently 153 engineering link capacity 161 engineering worst case 154 IP address 36 LAN engineering examples 157 MCDN over VoIP 105 147 MCDN Zone ID for SRG 144 multi locations
67. status settings on page 57 P0609327 02 73 Chapter 4 Installing NetVision telephones This section describes how to configure the Symbol NetVision handsets to the Business Communications Manager system Programming Note If your system is running DHCP ensure that you create a static IP list for all the NetVision telephones you want to program The information in this section includes e NetVision connectivity e Configuring NetVision records on page 75 e Modifying H 323 terminal records on page 81 NetVision connectivity The Business Communications Manager supports access points NetVision handsets and other wireless IP devices that use either IEEE 802 11 1 or 2 M bits sec Frequency Hopping Spread Spectrum or IEEE 802 11B 11 M bits sec Direct Sequence Spread Spectrum technology NetVision telephones use an enhanced version of H 323 NetVision and NetVision Data wireless IP telephones connect to the Business Communications Manager over a LAN through the Business Communications Manager LAN or WAN card The Business Communication Manager sees these telephones as IP telephones which means that the DN records are assigned from the digital range rather than from the Companion or ISDN range of DNs From within the system the handsets can make and receive calls from any trunk type supported by the system which can include voice over IP VoIP H 323 trunks digital and analog trunks The handset DN record
68. telephones You use the directional arrows on the telephone to access menu items which when selected perform as if you had entered that feature code This menu can also be accessed through the Services button default FEATURE 999 hot desking This feature allows you to transfer the telephone and call features temporarily from one IP telephone to another The originating IP telephone cannot be used during this period P0609327 02 Glossary 205 feature labels The names that appear beside the four six soft keys on Nortel IP telephones can be adjusted to better reflect local requirements Label changes are performed through the Unified Manager firewalls Firewalls are server security devices on a network that block or allow IP traffic into node networks or devices When configuring IP telephony you need to ensure that the port settings are correctly configured to pass through any network firewalls between the telephone and the Business Communications Manager full duplex transmission Simultaneous two way separate transmission in both directions G 711 For IP telephones the Business Communications Manager supports both a law and mu law variants of the G 711 CODEC 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 Choose the G 711 CODEC with the companding law alaw or ulaw that matches your system requirements Codecs with VAD
69. the published IP address of the Business Communications Manager see Finding the published IP address on page 35 IP Telephony Configuration Guide 50 Chapter 3 Installing IP telephones Table 12 IP telephone server configurations Continued Field Value Description If DHCP 1 Manual Cfg Full 0 If you indicate DHCP for the telephone but you want to enter DHCP Partial 1 static IP addresses choose 1 Partial If you choose 0 Full the DHCP server will assign IP addresses that are not static If DHCP 0 or Partial S1 IP ip address This is the Published IP address of the first Business Communications Manager that you want to register the telephone to 1 PORT Default 7000 This is the port the telephone will use to access this Business Communications Manager 1 ACTION Default 1 S1 RETRY COUNT digits between 0 Set this to the number of times you want the telephone to retry and 255 the connection to the Business Communications Manager S2 IP ip address This is the Published IP address of the second Business Communications Manager that you want to register the telephone to It can also be the same as the S1 setting S2 PORT Default 7000 This is the port the telephone will use to access this Business Communications Manager S2 ACTION Default 1 S2 RETRY COUNT digits between 0 and 255 Set this to the number of times you want the telep
70. to access features on the Business Communications Manager system and provides some quick tips for basic call functions The Business Communications Manager Telephony Features Handbook provides information about how to use Business Communications Manager call features The Business Communications Manager NetVision Phone Administrator Guide provides instructions for assigning features to the display list and includes an appendix containing a list of the features that work with NetVision handsets Codecs You can specify a preferred codec for your H 323 terminals As well you can set a specific codec in individual handset records This allows you to create the preferred call environment for your NetVision based communications The settings you choose depend on the expected network connection points of the handsets If all the handsets are expected to be used within a common and consistent network you can use the general default setting However some handsets that connect through busy systems may need specific settings to ensure consistent voice quality The default codec for NetVision handsets is G 729 However if the NetVision handsets connect over IP trunks the codec of the IP trunk takes precedence Caution NetVision handsets experience communications problems if your system has a NAT between the handset internet connection and the published address of the Business Communications Manager LAN For this reason this configuration is NOT s
71. trunks by adding the Remote Gateway information to the configuration On any IP gateway for which you want to allow fallback you need to ensure that QoS monitor is enabled Refer to Configuring a remote gateway H 323 trunks on page 103 and Configuring remote endpoints SIP trunks on page 106 The Remote Gateway screen is accessed at Services IP Telephony IP Trunks H 323 Trunks Remote Gateway and Services IP Telephony IP Trunks SIP Trunks Address Book Warning QoS monitor must be turned on at both endpoints QoS Monitor is found under Services For information about using the QoS monitor refer to Quality of Service Monitor on page 127 Network note All systems on your private network must be running BCM 3 5 or newer software or have the QoS patch installed that supports this release Business Communications Managers running BCM 3 0 1 or earlier software cannot provide a compatible VoIP trunk with BCM 3 5 or newer software without this patch P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 113 Describing a fallback network The following figure shows how a fallback network would be set up between two sites Figure 33 PSTN fallback diagram Business Communications Business Manager B Communications Manager A Public or Private Public or Private PSTN line PSTN line In a network configured for PSTN fallback there are two connections between a Business Communi
72. type of trunk in the same way that digital telephones do Before setting up IP clients you must enable keycodes for IP telephony For information about entering IP Client keycodes see the Keycode Installation Guide Each IP Client keycode opens a specific number of IP telephone channels on the system Channels on the MSC are distributed on a one to one basis as each IP telephone or NetVision handset registers with the system Detailed information about installing 120X X IP telephones is contained under the following headings e IP telephony on the Business Communications Manager on page 42 e Configuring Nortel Networks i series telephones on page 42 e Configuring DHCP on page 53 e Modifying IP telephone status settings on page 57 e Working with the features list on page 59 e Resetting the Hot Desking password on page 61 Download firmware to a Nortel IP telephone on page 65 e Deregistering DNs for IP telephones on page 66 e Moving IP telephones on page 69 e Configuring a new time zone on a remote telephone on page 70 e Nortel Networks 12050 Software Phone on page 71 IP Telephony Configuration Guide 42 Chapter 3 Installing IP telephones IP telephony on the Business Communications Manager The Business Communications Manager supports IP telephony protocols UNISTIM and H 323 version 4 e The Nortel Networks i series telephones use the UNISTIM protocol e The Symbol NetVis
73. user feedback IP Telephony Configuration Guide 198 Quality of Service Implementing QoS in IP networks The information under the headings in this section explain how to implement QoS in IP networks e Traffic mix on page 198 Business Communications Manager router QoS support on page 199 Corporate intranets are developed to support data services Accordingly normal intranets are designed to support a set of QoS objectives dictated by these data services When an intranet takes on a real time service users of that service set additional QoS objectives in the intranet Some of the targets can be less controlled compared with the targets set by current services while other targets are more controlled For intranets not exposed to real time services in the past but which now need to deliver IP telephony traffic QoS objectives for delay can set an additional design restriction on the intranet One method of determining requirements is to subject all intranet traffic to additional QoS restrictions and design the network to the strictest QoS objectives An exact plan for the design improves the quality of data services although most applications cannot identify a reduction of say 50 ms in delay Improvement of the network results in a network that is correctly planned for voice but over planned for data services Another plan is to consider using QoS in the intranet This provides a more cost effective solution to
74. 000 as a gatekeeper on page 136 for specific information about configuring the gatekeeper for H 323 trunks Network note If your private network contains a Meridian 1 IPT you cannot use Radvision for a gatekeeper Call Signaling Direct GateKeeperRouted GateKeeperResolved GatekeeperRoutesNoRAS Direct call signaling information is passed directly between endpoints The remote gateway table in the Unified Manager defines a destination code digits for each remote system to direct the calls for that system to route In each system the Nortel IP Terminals and H 323 Terminals records map IP addresses to specific telephones GatekeeperResolved all call signaling occurs directly between H 323 endpoints This means that the gatekeeper resolves the phone numbers into IP addresses but the gatekeeper is not involved in call signaling GatekeeperRouted uses a gatekeeper for call setup and control In this method call signaling is directed through the gatekeeper GatekeeperRoutedNoRAS Use this setting for a NetCentrex gatekeeper With this setting the system routes all calls through the gatekeeper but does not use any of the gatekeeper Registration and Admission Services RAS IP Telephony Configuration Guide 94 Chapter5 Configuring local VoIP trunks Table 23 Local Gateway IP interface fields Continued Field Value Description Gatekeeper IP lt ip address gt If GateKeeperRout
75. 1 Type a number between 001 and 999 This route defines the PSTN route to the other system Only numbers not otherwise assigned will be allowed by the system 2 Click the Save button Add the PSTN route to the local PSTN lines 1 Inthe Route field type a number between 001 and 999 This route defines the PSTN route to your local PSTN 2 Click the Save button Add the VoIP route 1 Inthe Route field type a number between 001 and 999 This route defines the VoIP route 2 Click the Save button Assigning the line pools to routes Assign the line pools to the routes you created in the previous section Figure 35 Route XXX screen Special International PSTN line pool to other system 1 On the navigation tree click the route you created for the PSTN line to the other system 2 In the Use Pool box type the letter of the line pool for the PSTN lines to the other system 3 In the External field If this is a public PSTN line enter the dial numbers that access the other system through the PSTN For example 1 area code gt lt local code 4 Inthe DN type box choose Public PSTN line pool to local PSTN lines 1 On the navigation tree click the route you created for your local PSTN line 2 Inthe Use Pool box type the letter of the line pool for the PSTN line IP Telephony Configuration Guide 116 Chapter6 Setting up VoIP trunks for outgoing calls 3 Inthe External field
76. 197 Decision does the intranet meet IP telephony QoS needs The end of the measurement and analysis is a good indicator of whether the corporate intranet can deliver acceptable voice and fax services The Expected QoS level column in the table indicates to the installer or administrator the QoS level for each site pair with the data Repeat this for each site pair At the end of the measurements the results are as shown in the following table Table 54 Expected QoS level per site Measured one way Measured packet loss delay ms Expected QoS level Destination pair Mean Mean o Mean Mean o Mean Mean o Santa Clara Richardson Good Good Santa Clara Ottawa Santa Clara Tokyo Richardson Ottawa Richardson Tokyo Ottawa Tokyo To provide voice and fax services over the intranet keep the network within a Good or Excellent QoS level at the Mean o operating area Fax services must not travel on routes that have Fair or Poor QoS levels If QoS levels of some or all routes fall short of being Good evaluate options and costs for upgrading the intranet The evaluation often requires a link upgrade a topology change or implementation of QoS in the network To maintain costs you can accept a Fair QoS level for the time for a selected route A calculated trade off in quality requires the installer or administrator to monitor the QoS level reset needs with the end users and respond to
77. 3 e Resource assessment on page 33 e Keycodes on page 34 e System configuration for IP functions on page 34 e Finding the published IP address on page 35 Media gateway parameters for IP service on page 37 e VoIP trunks on page 39 e TP telephone records on page 40 Network diagram To aid in installation a Network Diagram provides a basic understanding of how the network is configured Before you install IP functionality create a network diagram that captures all of the information described in the following table If you are configuring IP telephones but not voice over IP VoIP trunks you do not need to answer the last questions 1 d or 1 e Table 1 Network diagram prerequisites Prerequisites Yes 1 a Has a network diagram been developed 1 b Does the network diagram contain any routers switches or bridges with corresponding IP addresses and bandwidth values for WAN or LAN links Also refer to Appendix D Interoperability on page 175 1 c Does the network diagram contain IP Addresses netmasks and network locations of all Business Communications Managers 1 d Answer this if your system will use IP trunks otherwise leave it blank Does the network diagram contain IP Addresses and netmasks of any other VoIP gateways that you need to connect to IP Telephony Configuration Guide 32 Chapter2 Prerequisites checklist Table 1 Network diagram prerequisites Cont
78. 30 G 711 Payload Size ms 30 o lt vj Figure 24 SIP Media Parameters dialog box 9 Q IP Telephony System Configuration IP Terminals 9 IP Trunks H 323 Trunks e BIPTI nks PortRanges Media Parameters 1st Preferred Codec 6723 v 2nd Preferred Codec G7zluaw 3rd Preferred Codec G 711 aLaw v 4th Preferred Codec a73 vj Silence Compression Disabled Jitter Buffer Voice auro y IP Telephony Configuration Guide 90 Chapter5 Configuring local VoIP trunks 4 Use the information in the table below to set up the media parameters for your system Table 22 Media parameters record Field Value Description 1st Preferred Codec None Select the Codecs in the order in which you want the 2nd Preferred Codec G 711 uLaw system to attempt to use them 3rd Preferred Codec G 711 aLaw 4th Preferred Codec G 729 1st Preferred Codec None aioe rd Prelenediconee on G 729 VAD n rereme odec G 723 VAD 3rd Preferred Codec 4th Preferred Codec G 711 aLaw Performance note Codecs on all networked Business Communications Managers must be consistent to ensure that interacting features such as Transfer and Conference work correctly Systems running BCM 3 5 or newer software allow codec negotiation and renegotiation to accommodate inconsistencies in Codec settings over VoIP trunks Refer to Codecs on page 27 Silence Comp
79. 323ldentifier alphanumeric strings representing names e mail addresses etc Identified by the keyword NAME Example The Business Communications Manager is assigned a public dialed number prefix of 76 a private CDP number of 45 and an H323 Identifier alias Alias Names PUB 76 CDP 45 NAME bcm10 nortel com e H 225 Q 931 CallingPartyNumber NetCentrex gatekeeper The NetCentrex gatekeeper uses the H 225 Q 931 CallingPartyNumber to resolve the call originator for billing purposes This number must then contain a unique prefix or location code that is unique across all endpoints that are using the NetCentrex gatekeeper Identified by the keyword src Example for private networks CDP alias src lt DN gt UDP alias src lt LOC gt lt DN gt Example for public network src public OLI Note E164 or NPI TON alias types are commonly used since they fit into dialing plans A Business Communications Manager alias list should not mix these types Also the type of alias used should be consistent with the dialing plan configuration Use the same alias naming method on all Business Communications Managers within a network Registration TTL Default 60 seconds This TimeToLive parameter specifies the intervals when the VoIP gateway sends KeepAlive signals to the gatekeeper The gatekeeper can override this timer and send its own TimeToLive period P0609327 02 Chapter 5 Configuring local VoIP trunks 95
80. 4 recording routes 194 register IP telephone 43 IP telephones 51 Registration Disabled 51 Registration IP Radvision 135 Registration TTL Local Gateway 94 relocating IP telephones 69 Keep DN alive 69 remote access VoIP trunks 109 110 remote gateway configuring 103 106 destination digits 105 107 MCDN networking 147 network example 124 VoIP trunks 103 106 remote routers setting up 179 remote system VoIP trunks 85 resource assessment prerequisites 33 router Business Communications Manager QoS support 199 intranet resource considerations 162 IP telephones 49 links to virtual circuits 154 network prerequisites 31 number of hops 162 port settings 132 Traceroute 173 P0609327 02 Index 219 routes full duplex link 161 recording 194 site pairs 194 routing and hop count 163 asymmetrical 165 delay issues 165 instability 165 network example 125 PSTN fallback 118 VoIP trunks 114 S S1 Action 50 SIIP 50 S1 Port 50 S1 RETRY Count 50 S2 Action 50 S2IP 50 S2 Port 50 S2 RETRY Count 50 schedule activating VoIP schedule 119 control set 125 destination codes 117 PSTN fallback 118 service setting manual 118 SCNFallback 93 94 Scope status 53 send name display VoIP trunking 144 serial cable NetVision 75 serialization delay 163 SERVER NO PORTS LEFT 51 server parameters 49 SERVER UNREACHABLE RESTARTING 51 service setting manual 118 Services key feature 900 menu list 60
81. 421 to the gateway at 10 10 10 19 and the call is established If call signaling is set to Gatekeeper Routed and no pre granted ARQ has been issued 1 Business Communications Manager Ottawa sends an ARQ to the gatekeeper for DN 421 The gatekeeper resolves DN 421 to 10 10 10 17 Business Communications Manager Ottawa sends the call Setup message for DN 421 to the gatekeeper 10 10 10 17 which forwards it to the gateway at 10 10 10 19 4 The call is established IP Telephony Configuration Guide 142 Chapter 7 Optional VoIP trunk configurations Faxing over VoIP lines You can assign VoIP trunks to wired fax machines if you have T 38 fax enabled on the local gateway The Business Communications Manager supports this IP fax feature between Business Communications Managers running BCM 3 5 or newer software and between a Business Communications Manager running BCM 3 5 or newer software and a Meridian 1 running IPT 3 0 or newer software The system processes fax signals by initiating a voice call over the VoIP line When the T 38 fax packets are received at the remote gateway the receiving system establishes a new path that uses the T 38 protocol The remote gateway and any nodes between the two endpoints must be running BCM version 3 5 or newer software and have T 38 fax enabled on the system Refer to Setting up the local gateway on page 91 disrupt calls at other telephones using VoIP trunks in the vicinity of the fax
82. 51 relocating 69 restart to configure 48 review configuration information 51 router IP 49 server parameters 49 Set IP viewing 51 settings 57 slow connection 52 speech paths 52 staggered download 66 Troubleshooting 51 troubleshooting prompts 51 Unified Manager configuration 57 updating H 323 terminals record 81 VLAN service 40 VLAN settings 50 181 IP telephones see also 12001 12002 12004 12050 IP telephony asymmetrical media channel negotiation 178 Benefits 21 concepts 27 engineering link capacity 161 insufficient link capacity 162 Introduction 21 network checklist 31 network loading 160 network DHCP 53 networks 23 ongoing monitoring 165 setting QoS 191 WAN link resources 153 IP Terminal status 57 IP terminal status features list 59 IP trunking outgoing name display 144 IP trunks media resources 33 network prerequisites 31 IP TTL Traceroute 173 IP wireless keycode 74 IPT M1 protocol 25 IPWIs NetVision mode 80 J jitter 201 Jitter buffer adjust size 164 defined 28 IP telephones 46 Unified Manager settings 57 VoIP trunks 91 K Keep DN alive 69 keycodes IP telephones 41 NetVision 74 prerequisite list 34 VoIP trunks 86 LAN Business Communications Manager function 34 engineering examples 157 implementing the network 162 Published IP address 35 late packets 195 latency jitter buffer 46 line pool adding to DN record 109 network example 123 IP Telephony Configuration Gu
83. 56 LAN engineering examples lessen 157 WAN GOOIFeO eosessesenasmRre RR RS RET RRC HA RRG ER RS RR 158 QoS Monitoring Bandwidth Requirement 0 00 eee eee 159 Additional feature configuration woiscads aeu eu RT TRES TRRERTRE ER ERR TRES 159 Setting Non linear processing llle 159 Determining network loading caused by IP telephony traffic 160 ENOUGH INK Capeblly 2 iueud sure x mede m Rem ed R ER Rae eR bee amen tas 161 Nor enough nk CapDSIDy unu caras ke hee e RA ce ll C RC ee d 162 Other intranet resource considerations 00 cece eee eee eee 162 Implementing the network LAN engineering 0 0 2 eee eae 162 Farhgr nel galea 24 ue pepe ninr nitir URN RFEU RERDES REUTERS d P 162 Components of delay ci ced ele thay eel dee qme TAU AER RE EAR DR ERGO 163 uc M qe dT 163 Reducing HOP COUNT auus seek kk RR EE cc o 164 Adjust the jitter buffer size cissccas ska RR ty bedi CR R RR owes 164 PESO pco SN OIS oiacopequedesewgeue tq cesselesd qeu eub se ote wees 164 Roung SUGS M UUm 165 Post installation network measurements 0000 e eee eese 165 IP Telephony Configuration Guide 8 Contents Appendix B Sience compression seror siinon aha SERERE AERE DaeEREE E RETE 167 Silence compression on half duplexlinks lsleleeleeeleene 168 Silence compression on full duplex links llle 170 CORTOT NOSE PP
84. 729A is enabled silence suppression is disabled G 729AB and G 723 are disabled G 723 Not supported on CSE 1000 silence suppression is enabled G 723 G 723 is enabled silence suppression is disabled G 711 ulaw or G 711 alaw silence suppression has no effect G 729A and G 729AB are disabled G 711 is always part of the CSE 1000 configuration and cannot be removed Setting Codecs on the CSE 1000 Use the Element Manager tool to set the codec information for the CSE 1000 This tool can be accessed at http lt SignalingServerIP gt 1 Inthe tool select Configuration Select IP Telephony Click DSP Profile 2 3 Inthe Node Summary Window select the node to be configured and click Edit 4 5 On the list of codecs enable or disable each by clicking on the check box beside the codec name 6 To view or change the codec configuration click the codec name IP Telephony Configuration Guide 140 Chapter 7 Optional VoIP trunk configurations 7 Ensure the following fields are set Table 37 CSE 1000 codec configuration Field Value Description Codec Name codec name Name of the codec you selected Voice Payload Size msec per frame Choose the payload size for the codec Use 30 ms for interoperability with the Business Communications Manager Voice Playout Jitter Buffer digits Choose the minimum jitter buffer value you Nominal Delay want to allow Voi
85. 97 Setting up SIP trunk subdomain names You can specify the sub domain names associated with specific system dialing protocols for SIP call direction Domain names are used by SIP programming to determine the type of call being sent over the SIP trunk and where it needs to be routed Refer also to Configuring remote endpoints SIP trunks on page 106 1 Inthe Unified Manager click the keys beside Services IP Telephony and IP Trunking 2 Click SIP Trunks 3 Click on the Dialing Sub Domain tab Refer to the figure below Figure 27 SIP Dialing Sub Domain settings Dialing Sub Domain 2 164 National tional 164 e 164 Subscriber fo bscriber 164 2 164 Special special 164 2 164 Unknown unknown 154 Private UDP UDP Private CDP opp Private Special specialprivate Private Unknown uknownpivate Unknown Unknown unknow 4 Ifyou change any of the default settings ensure that you notify the system administrators for any systems with SIP trunks pointing to your system 5 When you are finished click anywhere on the navigation tree to exit and to commit the changes IP Telephony Configuration Guide 98 Chapter5 Configuring local VoIP trunks Viewing SIP summary and status SIP trunk programming provides a summary page that provides general information about the trunks on the system As well it indicates the current status of the trunks 1 In Unified Manager
86. Groper sends an ICMP Internet Control Message Protocol echo request message to a host It also expects an ICMP echo reply which allows for the measurement of a round trip time to a selected host By sending repeated ICMP echo request messages percent packet loss for a route can be measured Traceroute Traceroute uses the IP TTL time to live field to determine router hops to a specific IP address A router must not forward an IP packet with a TTL field of 0 or 1 Instead a router discards the packet and returns to the originating IP address an ICMP time exceeded message Traceroute sends an IP datagram with a TTL of 1 to the selected destination host The first router to handle the datagram sends back a time exceeded message This message identifies the first router on the route Then Traceroute transmits a datagram with a TTL of 2 Following the second router on the route returns a time exceeded message until all hops are identified The Traceroute IP datagram has a UDP Port number not likely to be in use at the destination normally gt 30 000 The destination returns a port unreachable ICMP packet The destination host is identified Traceroute is used to measure round trip times to all hops along a route identifying bottlenecks in the network 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
87. H 323 Endpoint Dialing Plan 00 000 197 Committing Gatekeeper Configuration Changes ssssassaa sasssa ana 138 Configuring Codec Compatibility llle 139 P0609327 02 Contents 7 salir Codecs on Te CSE 1000 2iesiuege Es e RERTEBETG Ne EE e4 eS 139 Gatekeeper call scenarios 00 000 eee 141 Faang over VolP INES PER C C E 142 Operational notes and restrictions lille 142 IP trunking interoperability settings lel ree 143 Configuring NetMeeting clients iuuclsaescuadceki suem de ka ackeloR ie XC ae 145 Chapter 8 Typical network applications using MCDN eeeesree 147 Setting up MCDN over VoIP with fallback 0 0 02 cee eee eee 147 MCDN functionality on fallback PRI lines 0 0 0 0 eee ee ee 148 Networking multiple Business Communications Managers 5 149 Multi location chain with call center 024 sce eoo une suae dew eee kg ak 151 Business Communications Manager to remote IP telephones 152 Appendix A Eficient NetWork uauusson das Rad BRE deRQREM RE EE E dura ics 153 Determining the bandwidth requirements lllslelleelelssn 153 Determining WAN link resources llle 153 Exi WUNZAUON a aodio 9m P DRRRCR IO RUE Eon 300 RO Rn o RE oad oon at 153 ii udis 154 Bandwidth requirements on half duplex links csse 155 Bandwidth requirements on full duplex links llle 1
88. ION Operations note Fax tones that broadcast through a telephone speaker will disrupt calls at other telephones using VoIP trunks in the vicinity of the fax machine Here are some suggestions to minimize the possibility of your VoIP calls being dropped because of fax tone interference Locate fax machine away from other telephones Turn the speaker volume on the fax machine to the lowest level or off if that option is available G 729 Payload Size ms 10 20 30 40 50 60 Default 30 G 723 Payload Size ms 30 G 711 Payload Size ms 10 20 30 40 50 60 Default 30 Set the maximum required payload size per codec for the VoIP calls sent over H 323 trunks Note Payload size can also be set for Nortel IP telephones Refer to Setting IP terminal general settings on page 43 Setting up the local gateway The call signaling method used by the local gateway defines how the Business Communications Manager prefers call signaling information to be directed through VoIP trunks Call signaling establishes and disconnects a call You set this information in the local gateway screens Modifying local gateway settings for H 323 and SIP trunks If the network has a gatekeeper H 323 trunks only 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 decid
89. IP sched Assign Normal sched under under destination code destination code First route XXA Use route XXB Absorb length 1 Absorb length 1 System dialout to remote system _ If fallback occurs gs System grabs PSTN line pool and dials out System grabs VoIP pool lt public to remote gt 2233 and dials out 2233 P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 121 Figure 39 Setting up routes and fallback for remote external call CDP dialing code Note For this example the destination code to call to the PSTN attached to the other system is 9 lt areacode gt Both systems have destination code 9 set up as the local PSTN access code Call to local PSTN connected to remote BCM system User dials 9 lt areacode gt lt local PSTN gt Example 92045554678 Create route to VoIP pool Create route to PSTN pool Route YYA VoIP pool Route YYB PSTN line pool No external External lt public toremote gt DN type Public DN type Public Destination code 9 schedules Assign VOIP sched under Assign Normal sched under desti destination code nation code First route YYA Use route YYB Absorb length 0 Absorb length 0 System dialout to remote system If fallback occurs System grabs the PSTN line pool and dials out lt public toremote gt 92045554678
90. In 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 e 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 e Delay variation jitter The amount of variation in packet delay is otherwise known as delay variations or jitter Jitter affects the ability of the receiving gateway to assemble voice packets received at irregular intervals into a continuous voice stream Fallback to PSTN If the measured Mean Opinion Score MOS for all codecs is below the configured threshold for any monitored gateway the Fallback to PSTN activates This feature 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 bad network conditions remote gateway is out of service e no network connection e not en
91. M BCM CIR DID DOD DIBTS DSB DSL DSP FEPS FoIP FUMP ICMP TEEE802 ESS IP IPT ISP ITG ITU IXC IP ISDN Kb LAN LATA LEC Mb Application Programming Interface Asynchronous Transfer Mode Business Communications Manager Committed Information Rate Direct Inward Dialing Direct Outward Dialing Digital In Band Trunk Signaling DIBTS Signaling Buffer Digital Subscriber Line Digital Signal Processor Functional Endpoint Proxy Server Fax over IP Functional Messaging Protocol Internet Control Message Protocol Institute of Electrical and Electronics Engineers Inc standard 802 Electronic Switching System Identification code Internet protocol Internet Protocol for Telephony for Meridian supported by BCM version 3 5 and newer software Internet Service Provider Internet Telephony Gateway for Meridian supported by BCM version 3 0 1 and earlier software providing the systems do not share a network with a BCM version 3 5 or newer software International Telecommunication Union IntereXchange Carrier Internet Protocol Integrated Services Digital Network kilobit kilo Byte Local Area Network Local Access and Transport Area Local Exchange Carrier Mega bit IP Telephony Configuration Guide 18 Preface MB MOS NAT NVPA PCM PING PiPP PPP PRI PSTN QoS RAS RTP SIP SNMP TCP UDP UTPS VoIP VAD VLAN WAN Mega Byte Mean Opinion Score Network Address Translation NetVision Pho
92. P Telephony rLocal Gateway IP Interface System Configuration IP Terminals 9 IP Trunks Fallback to Circuit Switched Enabled All v H 323 Trunks SIP Domain yourcompany com 9 SIP Trunks Address Book Transport upP P0609327 02 Chapter 5 Configuring local VoIP trunks 93 3 Use the information in the table below to set up the Local Gateway IP interface record Table 23 Local Gateway IP interface fields Field Value Description Fields that appear for both types of trunks Fallback to Circuit Switched Enabled All Enabled TDM only Disabled Your choice determines how the system will handle calls if the IP network cannot be used Enabled All All calls will be rerouted over specified TDM trunks lines Enabled TDM only All voice calls will be rerouted over specified TDM trunks lines Disabled Calls will not be rerouted Call Signaling Poean Enabled TDM only Gatekeeper IP Disabled Note Enabled TDM only enables fallback for calls originating on digital telephones This is useful if your IP telephones are connected remotely on the public side of the Business Communications Manager network because PSTN fallback is unlikely to result in better quality of service in that scenario Fields that appear only for H 323 trunks Configuration note Refer to Using Radvision ECS 3 2 GK as the gatekeeper on page 134 and Using CSE 1
93. P Terminals The IP Terminal summary appears 2 Click the Telephony Features list tab Figure 9 IP Telephony Features List Summary General IP Terminal Status Telephony Features List Telephony Features List Feauelndex Feature Name FeatueCode F1 Hot desking F2 Last Number Redial F3 Conference F4 Do Not Disturb F5 Call Forward F6 Page F Background Music F8 Call Park F3 Call Pickup Voice Call Speed Dial Message Send SWCA 1 3 Select the feature you want to modify and right click or click the Configuration menu item then select the action you want to perform Performance Faul Modify parameters Deregister DN Force firmware download Reset Hot Desking Password Modify Feature Delete Feature The Telephony Features list screen appears IP Telephony Configuration Guide 60 Chapter 3 Installing IP telephones Figure 10 Add Modify Telephony Features List Telephony Features List Feature Index Fe Feature Name SwA 1 Feature Code Format 10122456783 4 Enter or change the Feature Name and corresponding Feature Code in the appropriate fields 5 Click the Save button The features list appears Notice that the system assigns a Feature Index number adding the feature to the bottom of the list Refer to the information under Using the Services button to access features for a description about how t
94. P Trunks PortRanges The IP Terminal summary screen appears Figure 6 IP terminal registration server status Configuration Performance Fault Report Tools Logoff View Help Status Up hd Version 30 100 90 12 Description UNISTIM Terminal Proxy Server 2 The following fields provide information about the IP server Only the status field is configurable Table 15 IP terminal Summary fields Field Value Description Name UTPS Name of the server Status Up UP server is operating Enabled Enabled Server is using DHCP Disabled Disabled server is not working Version read only current version of server software Description read only description of server P0609327 02 Chapter 3 Installing IP telephones 57 Modifying IP telephone status settings Settings such as jitter buffers and codecs for the Nortel IP telephones can be modified through the Unified Manager 1 Inthe Unified Manager open Services IP Telephony IP Terminals and click Nortel IP Terminals The IP Terminal summary appears 2 Click the IP Terminal Status tab On the IP Terminal status screen every IP telephone currently connected to the Business Communications Manager occupies a row in the IP Terminal Status table as shown in the figure below Figure 7 IP Terminal status cdit Configuration Performance Fault Report Tools Logoff View Help IP Terminal Status Telephony Features List P T
95. PSON us Lue RUE APR p OU eg dba pola d qd NP RR wae 178 Supported voice payload sizes llle 183 Software interoperability restrictions and limitations for IP trunking 184 Software network communications application compatibility 188 1 38 restrictions and req iremenis sees ka Rm a onde gm aac Rma Rn 189 Quality of valeg Service ues Rede d VROUE ROSE eee RR A GC XR RE RR 192 Sile pairs and POUL uL irure XE RUE keit c o deed Rawk e Ep do aar keds 194 Computed load of voice traffic per link llle 195 Delay and error statisti S 2 0c ccnedeaedeiae eee ERE Gp x ER b EY ERA 196 Expected QoS level par SII sn iiuaacb specia end DEPENEN ENPE d a ecd 197 P0609327 02 15 Preface This guide describes IP Telephony functionality for the Business Communications Manager system that is running BCM 3 6 software This information includes configuration instructions for Nortel IP telephones i series 200X the 12050 Software Phone the Symbol NetVision and NetVision data telephones H 323 protocol devices and VoIP trunks H 323 and SIP as well as some general information about IP networking data controls and IP private telephony networking Before you begin This guide is intended for installers and managers of a Business Communications Manager system Prior knowledge of IP networks is required Before using this guide the Business Communications Manager system must be configured and tested for bas
96. Part No P0609327 02 March 17 2004 Business Communications Manager 3 6 IP Telephony Configuration Guide TEL NETWORKS Copyright 2004 Nortel Networks All rights reserved The information in this document is subject to change without notice The statements configurations technical data and recommendations in this document are believed to be accurate and reliable but are presented without express or implied warranty Users must take full responsibility for their applications of any products specified in this document The information in this document is proprietary to Nortel Networks NA Inc Trademarks NORTEL NETWORKS is a trademark of Nortel Networks Microsoft MS MS DOS Windows and Windows NT are registered trademarks of Microsoft Corporation Symbol Spectrum24 and NetVision are registered trademarks of Symbol Technologies Inc All other trademarks and registered trademarks are the property of their respective owners P0609327 02 Contents ar e rrrrTPTTT 15 Pere VOU DOCE 4 115916 dcc hie dett Bo Bo Bd o de Bo RE E Rode Bie a oes d old 15 Symbols used I TE QUI ous mar E RER E REUS FREESDPCTAFRBAS PTT E REA RR RE s 15 Wert COVOFBIRS 4 3 9943 QR MEC X eC S CENSOR EXER RCR RACER EA E RC Ro EA 16 Pol 1 i eo ere d donee Ed disegno x idiot oda tru bdo o x oci dia apps 17 Related publications lt 0 24456 24s0se eer ceierieeareeeeeRnee eee eos meh de god 18 How te get Help 4a cessar
97. RI IP Telephony Configuration Guide 218 Index using MI IPT 25 PRI MCDN fallback 148 private IP address 32 36 122 private network MCDN Zone ID 144 private network virtualID 144 prompts IP telephones configuration 51 propagation delay 163 protocol link bandwidth requirements 155 156 remote gateway 103 PSTN fallback 112 activating VoIP schedule 119 configuring 112 destination codes 116 dialed digits 114 MCDN networking 148 mean opinion score 201 PRI line 123 scheduling 118 public IP address 32 36 122 Published IP address choosing 36 determine which IP address to use 36 IP telephones 50 network example 123 setting 35 VoIP trunks 35 QoS analysing 197 Business Communications Manager gateway router support 199 defined 29 implementing in IP networks 198 MCDN networking 148 measuring intranet 193 MOS range qualitative scale 192 objectives 191 parameters 154 setting 191 status 127 QoS monitor enabled 124 remote gateway 103 status display 127 updating data 128 qualitative scale QoS 192 Quality of Service Monitor see QoS monitor 127 queuing delay 163 R RI determining link capacity 161 peak VoIP load 161 R2 determining link capacity 161 peak VoIP load 161 Radivision interoperability support 135 Radvision ECS 2 1 0 1 gatekeeper 134 mandatory fields 95 Predefined Endpoints Properties settings 135 receive fallback threshold 192 receive path 156 receive threshold 103 12
98. S Telephone European Freephone 00800 800 89009 European Alternative United Kingdom 44 0 870 907 9009 Africa 27 11 808 4000 Israel 800 945 9779 Note Calls are not free from all countries in Europe Middle East or Africa Fax 44 191 555 7980 email emeahelp nortelnetworks com CALA Caribbean amp Latin America Technical Support CTAS Telephone 1 954 858 7777 email csrmgmt nortelnetworks com APAC Asia Pacific Technical Support CTAS Telephone 61 2 870 8800 Fax 61 388664644 email asia support nortelnetworks com In country toll free numbers Australia 1800NORTEL 1800 667 835 China 010 6510 7770 IP Telephony Configuration Guide 20 Preface India 011 5154 2210 Indonesia 0018 036 1004 Japan 0120 332 533 Malaysia 1800 805 380 New Zealand 0800 449 716 Philippines 1800 1611 0063 Singapore 800 616 2004 South Korea 0079 8611 2001 Taiwan 0800 8 10 500 Thailand 001 800 61 1 3007 Service Business Centre amp Pre Sales Help Desk 61 2 8870 5511 P0609327 02 21 Chapter 1 Introduction IP telephony provides the flexibility affordability and expandability of the Internet to the world of voice communications This section includes an overview of the components that make up the Business Communications Manager version 3 6 IP telephony and Voice over IP VoIP features e IP telephones and VoIP trunks on page 22 e Creating the IP telephony network on page 23 e Key IP tele
99. Santa Clara Business Communications Manager e 3322is linked to the target line associated with DN 3322 e The call arrives at the phone with the DN 3322 If a user in Santa Clara wanted to make a local call in Ottawa they would dial 29 followed by the local Ottawa number The digit 2 accesses the remote gateway for the VoIP line The digit 9 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 1200X telephone on the system For information about configuring 1200X telephones see Chapter 3 Installing IP telephones on page 41 Note IP telephones require an IP network to reach the Business Communications Manager However they do not need to use VoIP trunks to communicate beyond the Business Communications Manager They can use any type of trunk IP Telephony Configuration Guide 126 Chapter6 Setting up VoIP trunks for outgoing calls In this case the Santa Clara administrator wants to connect an i2004 phone using the LAN 1 network interface 1 The installer sets up the Business Communications Manager to handle the IP telephone by turning Registration to ON and Auto Assign DNs to ON 2 The installer connects the telephone to the LAN and sets it up using the following settings e Set IP address 10 10 5 10 e Default GW 10 10 5 1 This is the IP address of the default gateway on the network which is the nearest
100. The following figure shows components of a Business Communications Manager network configuration In this example two Business Communications Manager systems are connected both through a PSTN connection and through a WAN connection The WAN connection uses VoIP trunks If the PSTN connections use dedicated ISDN lines the two systems have backup private networks to each other Both Business Communications Manager systems use VoIP trunks through a common WAN to connect to the Meridian M1 IPT system IP Telephony Configuration Guide 24 Chapter1 Introduction Figure 1 Network diagram Business Communications Manager A Router IP telaphon A Digital telephone A NetVision H 323 device A 12050 telephone A Router Gatekeeper ED Business Communications Manager B M1 IPT n Meridian set A Networking with Business Communications Manager The Business Communications Manager is a key building block in creating your communications network It interoperates with many devices including the Meridian 1 system and H 323 devices The Business Communications Manager system can be connected to devices through multiple IP networks as well as through the PSTN Multiple Business Communications Manager systems also can be linked together on a network of VoIP trunks and or dedicated physical lines Refer to Chapter 8 Typical network applications using MCDN on page 147 Th
101. UDP port 5000 If you use QoS Monitoring in your gateway setting please refer to the following paragraph for a description of bandwidth requirement of QoS Monitoring There are a total of 25 monitoring packets traveling in each direction every 15 seconds Each of monitoring packages has 88 bytes in IP layer These monitoring packets are equally spaced out in the 15 second intervals For example if there are two Business Communications Managers BCM A and BCM B connected to each other with QoS Monitoring enabled then in every 15 seconds there are 25 monitoring packets going from BCM A to BCM B and then back to BCM A Similarly 25 packets go from BCM B to BCM A then back to BCM B In other words in this case the overhead in IP layer caused by these monitoring packets is about 2x25x88 15 293 bytes second in one direction Additional feature configuration This section contains additional information about configuring your network to run efficiently Refer to the information under the following headings for details e Setting Non linear processing Determining network loading caused by IP telephony traffic on page 160 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 H 323 settings The H 323 parameters appear in the right window 2 From the Non linear processing menu select either Enabled or Disabled
102. VoIP call Gatekeeper note If your system is controlled by a gatekeeper you do not need to establish these gateways Refer to Using a gatekeeper on page 133 Creating a remote gateway record To add an entry to the H 323 trunk remote gateway list 1 InUnified Manager click the keys beside Services IP Telephony IP Trunks H 323 Trunks 2 Click Remote Gateway The remote gateway tab appears The Remote Gateway screen shows all gateway records that have been added to the system 9 Q IP Telephony System Configuration IP Terminals 9 Q IP Trunks 9 H 323 Trunks E SIP Trunks PortRanges 3 On the top menu click Configuration and select Add entry If you are modifying an existing entry select the entry on the Remote Gateway screen then under Configuration select Modify entry Modify Entry Add Entry Delete Entry 4 The Remote Gateway dialog box appears as shown in the next figure IP Telephony Configuration Guide 104 Chapter6 Setting up VoIP trunks for outgoing calls Figure 30 Remote gateway dialog box Remote Gateway Name 0000 Destination IP 555 506 000 000 QoS Monitor Disabled v Transmit Threshold o Receive Threshold o Gateway Type BCM3 5 v Gateway Protocol None v Format 15 characters Destination Digits 0000 5 Use the information in the table below to set up the remote gateway information Table 25 Remote gateway record
103. a processed as output by a computer communications device link or system topology Logical or physical arrangement of nodes or stations Traceroute Traceroute uses the IP TTL time to live field to determine router hops to a specific IP address UNISTIM Terminal Proxy Server UTPS This is a Nortel designed protocol for IP telephony applications The Nortel IP telephones for instance use this protocol to communicate with the Business Communications Manager IP Telephony Configuration Guide 210 Glossary voice compression Method of reducing bandwidth by reducing the number of bits required to transmit voice Voice over IP VoIP trunks VoIP trunks are virtual telephone lines that the Business Communications Manager uses instead of wired lines to transfer IP traffic to other compatible systems with VoIP trunks Both digital and IP telephones can use these channels The Business Communications Manager supports trunks using the H 323 and SIP protocols VoIP fax Wired fax devices can be assigned to H 323 VoIP line pools as these VoIP trunks now support the T 38 fax protocol P0609327 02 211 Index Numbers 3 port switch IP telephones 46 relocating IP telephones 69 A absorbed length 117 118 access code network example 122 acronyms 17 active calls deregistering disruption 66 Address Range IP telephones 53 a law 178 Alias Names Local Gateway 94 Aliases Radvision 135 Allowed Servic
104. addgud 34 Business Communications Manager system configuration 34 Published IP Address options uaa accord dice qnc x dana ic Ro dcos 35 IP terminals general record fields sso ken 37 VoIP trunk provisioning saci a dodo e eh YOU EROR OR GU AGNES ROG MUR 39 IP telephone provisioning iiss ecce ex xr Sew i dee dukke RACE d loco ees 40 IP terminals general record fields llli 44 IP telephone server configurations llle 49 IF telephony display messages ssuesdeccak ie RS REG Ge ER a RR ranas 51 IP telephone troubleshooting llllselesellleleleeee 52 IP terminal Summary fields uuu x ER x RR REC e ae aodio bd 56 IP Terminal Status fields os adn xac edic gadAmQTCERPEGGeERREPREFXZASR PASE 58 Relabelling examples 1 4255 020054 4o5544 e mRbeekadeerbeeex erede 65 gcc NEE MecCC TP rM 76 H 323 terminals Summary fields llllllllllsllesn TE idee Ie PE 80 Media parameters record osossaascasauuee rata 0088S 04244 PRG EAE DORE 88 Media parameters record iosooeesecee mm nes Exe RR Re eS 90 Local Gateway IP interface fields iui scsi dua eae eae scelte chek 93 Route and Dialing Plan configurations for NPI TON sees 96 Remote gateway record iis se egqibeware Exdev Veber XS dade ys ad Us 104 Adding SIP Address Book records 00 cece eee eee 106 Fallback configuration for to create fallback between two systems 123 GOS SIME 65 264 nobis bu de nut SANDER LADS HEAD
105. al or extended LAN segment even though they may be geographically separated VLAN IDs are determined by how the VLAN switch is configured If you are not the network administrator you will have to ask whoever manages the switch what the VLAN ID range is for your system Also refer to Choosing DHCP for VLAN and Specifying the site specific options for VLAN on page 182 VLANs aim to offer the following benefits e VLANs are supported over all IEEE 802 LAN MAC protocols and over shared media LANs as well as point to point LANs e VLANs facilitate easy administration of logical groups of stations that can communicate as if they were on the same LAN They also facilitate easier administration of move add and change in members of these groups e Traffic between VLANs is restricted Bridges forward unicast multicast and broadcast traffic only on LAN segments that serve the VLAN to which the traffic belongs e For IP telephony VLANs provide a useful technique to separate and prioritize the telephony traffic for L2 switches e VLAN also provide a shield from malicious traffic that may be targeted at the IP phone in order to steal or disrupt service e Reuse IP address in different VLANs e As far as possible VLANs maintain compatibility with existing bridges and end stations e fall bridge ports are configured to transmit and receive untagged frames bridges will work in plug and play ISO IEC 15802 3 mode End stations will
106. aling plan 124 private network MCDN 148 changes to the intranet 165 checklist 31 clients media resources voice mail media resources WAN media resources 33 codecs defined 27 first preferred codec 177 for IP telephones 45 handling on network 154 types bandwidth 154 Unified Manager settings 57 IP Telephony Configuration Guide 212 Index comfort noise 172 computed load 194 computer IP telephony prerequisites 40 Conference Call 177 configure DN record 51 12050 Software Phone 71 IP server parameters 49 restartto 48 review information 51 Connecting to Server 51 contrast level IP telephones 52 control set setting the schedule 125 conventions and symbols 15 text 16 Coordinated Dialing Plan see CDP 124 CSE MCDN for IPT 105 customize feature labels 63 D Danger symbol 15 Default gateway IP telephones 49 53 delay characteristics 194 end to end 163 gathering statistics 196 link 163 network analysis 162 propagation 163 queuing 163 routing and hop counts 163 serialization 163 deleting handset record 82 deregister IP telephones 66 destination codes for fallback 116 PSTN fallback 116 remote gateway destination digits 117 118 schedule 117 destination digits destination code 117 118 network example 124 remote gateway 103 destination gateway 172 destination IP network example 124 remote gateway 103 DHCP configuring 53 configuring for IP telephones 53 Invalid Server Address 54 IP telep
107. alysis 162 Sniffer 173 trunks VoIP 22 two way call bandwidth requirements 155 U UDP port 173 portranges 132 private network MCDN 148 Unified Manager deleting handset record 82 destination codes 116 DN record 109 H 323 Terminals record 79 H 323 Trunks record 89 103 106 setting up target lines 98 Unified Messaging 149 usable link bandwidth number of calls 155 V VAD silence suppression 167 Virtual Private Network ID 144 VLAN 50 IP telephone 50 181 i series telephones 40 site specific options 182 Voice Activity Detection VAD 167 177 Voice Activity Detection see VAD 167 voice compression 176 voice jitter buffer 91 voice mail VoIP trunk MWI interoperability 143 voice path silence suppression 156 voice quality codec 45 jitter buffer 46 VoIP DISA 85 gateway progress tones 178 gateway prerequisites 31 implementing QoS into network 198 load 161 MCDN network 105 schedule activating 119 schedule setting up 118 trunks configuring 85 VoIP trunks activating VoIP schedule 119 adding to DN records 109 configuration 85 configuring incoming calls 98 configuring NetMeeting clients 145 connecting IP telephones 125 defined 22 destination codes 116 destination digits 105 107 example configuration 122 global IP 35 incoming call configuration 98 jitter buffers 91 keycodes 86 making calls 125 media parameters 89 networking IP address 36 networking multiple systems 149 networking remo
108. ance for the feature P0609327 02 Chapter 3 Installing IP telephones 65 Some features like Page and System Wide Call Appearances SWCA have several variations of feature invocation that you may want to customize for the users Paging can be F60 F61x F62 and F63x System wide Call Appearance SWCA has 16 codes 521 to 536 The following table shows examples of changing labels for page codes and SWCA codes Table 17 Relabelling examples Feature code New label Feature code New label 60 Gen Page 521 SW Call 1 610 Pg Every 522 SW Call 2 61 Zone lt digit from 523 SW Call 3 1 9 gt 62 Speak Pg 524 SW Call 4 630 Speak All 525 SW Call 5 Note Line names are defined when you configure the line and can be changed through the Lines menus Download firmware to a Nortel IP telephone Firmware is the software stored in the telephone When the Business Communications Manager is upgraded with a new IP telephone firmware load this firmware load automatically downloads into the IP telephones when they next connect to the Business Communications Manager You can use the Force firmware download option under the Configuration menu Nortel IP Terminals to force immediate download to a telephone You would do this in situations where you suspect that a particular telephone has corrupted firmware Forcing a firmware download to an IP telephone Follow these steps to force a firmware download to a t
109. bels on the memory buttons for details about changing feature labels IP Telephony Configuration Guide 64 Chapter3 Installing IP telephones Changing features or labels on the memory buttons Follow these steps to change the features or labels on the memory buttons on your IP telephone 1 Click the Telephony Services General Nortel IP terminals and Feature labels keys 2 Click the label set you want to view The Labels abel number screen appears Figure 12 Label set defaults Labels 7312 rLabels 7 12 Feature 7 982 Label VMail oper Feature 8 984 Label 8 Cfwd vmail Feature 3 985 Label 3 VMail DN Feature 10 986 Feature settings oe Nortel IP terminals Label 10 xfer vmail Q Feature labels z Labels 1 6 Feature 11 987 Label 11 Mail intr Labels 13 18 i Labels 19 24 Feature 12 988 Label12 VMail dir 3 If you have an existing list or you do not want to change any defaults go to the first empty Feature field 4 Inthe Feature label number field enter the feature code for the feature you want to relabel Example enter 3 for conference call 5 In the Label label number field enter the new label you want the telephones to display Example The current label for feature code 3 is Conference you could change it to Conf Call 6 Click anywhere outside the field to save the changes The system automatically updates any IP telephones that have a button appear
110. bers of hunt groups but when they are answer DN twinned with other sets they do not ring under some circumstances When configured with an answer DN for a set in a hunt group Symbol sets sometimes do not ring or ring but do not display CLID information and cannot answer the incoming call It is recommended that the Symbol set be added to the hunt group before the answer DN set or that the Symbol set be designated as the prime DN with the answer DN for it applied to the twinned desk set This does address most of the functionality problems There still appears to be a problem for calls routed by CCR IP Telephony Configuration Guide 188 Interoperability The following table shows which networking applications are supported for each Business Communications Manager software release Table 48 Software network communications application compatibility Application compatibility BCM BCM 2 5 2 5 FP1 BCM 3 0 BCM Net ITG IPT v BCM version 2 03 2 5 FP1 MR1 3 0 1 3 5 Meeting X X Symbol GK CSE1K BCM 2 03 X basic call TG v to from 25 24 BCM 2 5 X X basic call TG v to from 25 25 BCM 2 5 FP1 X X X X ITG 25 25 X FP1 MR 1 1 X X X X ITG 25 25 X X BCM 3 0 X X X X ITG 26 26 X X X BCM 3 5 x x x x X X IPT 3 0 3 1 X X X BCM 3 6 x x x x X X IPT 3 0 3 1 X X X with QoS patch 3 0 0 25 or greater TG is not supported on a private network that
111. cations Manager QoS Monitor Enabled Transmit Threshold 3 5 moderate quality Receive Threshold 3 5 moderate quality Gateway Type BCM3 6 Gateway protocol None Destination Digits Destination Digits Remote Gateway Destination digits note In this case the systems use a Coordinated Dialing Plan CDP network and the destination digit is included in the DN Ottawa 2 Santa Clara 3 Set up Scheduling to allow Service setting Manual Services Telephony Services you to manually start and Overflow Y Scheduled Services Routing stop schedules Services VoIP Schedule 4 Confirm or set up a route Route 009 Services Telephony Services using the line pool to access the local PSTN External to Santa Clara 1408555 Line Pool lt publiclinepool gt DN type Public External to Ottawa 1613555 Call routing Routes Route 009 Set up a route that contains the PRI fallback lines Route 774 Dialout N A PSTN Line Pool PRI A DN type Private Services Telephony Services Call routing Routes Route XXX P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 125 Table 27 Fallback configuration for to create fallback between two systems Continued VoIP Line Pool O DN type Private Settings for Settings for Task Santa Clara Ottawa Location in Unified Manager Set up a route that contains Route 867 Services Telephony S
112. cations Manager and a remote system e One connection is a VoIP trunk connection through the IP network e The fallback line is a PSTN line which can be the public lines or a dedicated T1 BRI PRI or analog line E amp M to the other system When a user dials the destination code the system checks first to see if the connection between the two systems can support an appropriate level of QoS If it can the call proceeds as normal over the VoIP trunk If the minimum acceptable level of QoS is not met the call is routed over the second route through the PSTN line For PSTN fallback to work you must ensure that the digits the user dials will be the same regardless of whether the call is going over the VoIP trunk or the PSTN In many cases this involves configuring the system to add and or absorb digits This process is explained during the steps in Configuring routes for fallback on page 114 and Adding the destination code for the fallback route on page 116 For detailed information about inserting and absorbing digits see the Programming Operations Guide IP Telephony Configuration Guide 114 Chapter6 Setting up VoIP trunks for outgoing calls Configuring routes for fallback Configuring routes allows you to set up access to the VoIP and the PSTN line pools These routes can be assigned to destination codes The destination codes then are configured into schedules where the PSTN line is assigned to the Normal schedule and
113. ce Playout Jitter Buffer digits Choose the maximum jitter buffer value you Maximum Delay want to allow VAD checkbox enabled Check or uncheck box to enable or disable disabled silence suppression for the codec 8 Click Submit 9 Click Transfer for the node that you modified P0609327 02 Chapter 7 Optional VoIP trunk configurations 141 Gatekeeper call scenarios This section explains what must be set up and how a call would be processed for the two types of gatekeeper configurations The following figure shows a network with three Business Communications Managers and a gatekeeper Figure 46 Business Communications Manager systems with a gatekeeper gatekeeper IP 10 10 10 17 Business Communications Manager Ottawa DN 321 IP 10 10 10 18 Business Communications Manager Santa Clara IP 10 10 10 19 Business Communications Manager Calgary IP 10 10 10 20 This example explains how a call from DN 321 in Ottawa would be made to DN 421 in Santa Clara It assumes that call signaling is set to Gatekeeper Resolved and no pre granted AdmissionRequest ARQ has been issued 1 Business Communications Manager Ottawa sends an 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
114. changing the DN 1 a Ff o N If you want to retain DN specific features such as Call Forward No answer and Call Forward on Busy if an IP telephone becomes disconnected you must activate the Keep DN alive setting as described below Otherwise go to step 2 a Inthe Unified Manager under the Services Telephony Services list click the DN record for the IP telephone b Click the Capabilities heading C Beside the Keep DN alive field choose Y Choosing N for this field allows the DN record to become inactive if the IP telephone is disconnected This produces a Not in Service prompt if any of the special features such as Call Forward are invoked Warning If the system is reset while an IP telephone is disconnected the Keep DN alive feature becomes inactive until the telephone is reconnected This setting must be enabled if you want to change the codec or jitter buffer for an IP telephone that is offline Note When an IP telephone is disconnected there is about a 40 second delay before the system activates Keep DN alive during which incoming calls will either get a busy signal or be rerouted to the Prime set depending on how your system is programmed The same type of delay occurs when the IP telephone is reconnected to the system Disconnect the power from the IP telephone or 3 port switch Disconnect the network connection At the new location reconnect the network cable and the power connection If the new location is on a d
115. cific option in the DHCP offer message The following definition describes the Nortel IP telephone specific Site Specific option This option uses the reserved for site specific use DHCP options DHCP option values 128 to 254 and must be returned by the DHCP server as part of each DHCP OFFER and ACK message for the IP telephone to accept these messages as valid The IP telephone will pull the relevant information out of this option and use it to configure the IP phone Format of field is Type Length Data Type 1 octet Five choices 0x80 0x90 0x9d Oxbf Oxfb 128 144 157 191 251 Providing a choice of five types allows the IP telephone to work in environments where the initial choice may already be in use by a different vendor Pick only one TYPE byte Length 1 octet variable depends on the message content Data length octets e ASCII based format VLAN A XXX YYY ZZZ where VLAN A uniquely identifies this as the Nortel DHCP VLAN discovery A signifies this version of this spec Future enhancements could use B for example ASCII comma is used to separate fields ASCII period is used to signal end of structure XXX YYY and ZZZ are ASCII encoded decimal numbers with a range of 0 4095 The number is used to identify the VLAN Ids A maximum of 10 VLAN Ids can be configured NONE means no VLAN default VLAN The DHCP Offer message carrying VLAN information has no VLAN tag when it is sent o
116. codes you have installed on your system IP Telephony Configuration Guide 88 Chapter5 Configuring local VoIP trunks 4 Use the information in the table below to determine the distribution of H 323 and SIP trunks on your system Table 21 Media parameters record Field Value Description Maximum Trunks read only This value is the total number of VoIP trunks you can have on your system usually 60 Total Trunk Credits read only This value is determined by the number of VoIP trunk keycodes you have installed on your system 4 8 12 and so on Number of H 323 lt digits gt Enter the total number of H 323 trunks out of the total Trunks number of credits you have available Number of SIP Trunks lt digits gt Enter the total number of SIP trunks out of the total number of credits you have available The sum of these numbers must not exceed the Total Trunk Credits available Click anywhere on the navigation tree to exit this screen and activate the settings Go to Services Telephony Services Lines VoIP lines Enabled VoIP lines and configure the lines into line pools Change the other settings as you would for any other lines Refer to the Lines and Loops chapter in the Programming Operations Guide for details Go to Services System DNs Active Set DNs and select the DNs for the telephones that need access to these lines and add the VoIP line pool s to the DN record Refer to the chapt
117. configuration information the telephone attempts to connect to the Business Communications Manager The message Locating Server appears on the display If the connection is successful the message changes to Connecting to Server after about 15 seconds Initialization may take several minutes Do not disturb the telephone during this time When the telephone connects to the server and is ready to use the display shows the time and date As well the six keys at the top of the display are labelled If you experience problems with IP telephone registration refer to the section Troubleshooting IP telephones Notes e Ifthe DN record has not yet been configured as will be the case with auto assigned DNs you will only be able to make local calls until other lines have been assigned in the DN record e If the telephone has not been registered before you will receive a New Set message Enter the information as prompted Refer to Registering the telephone to the system on page 48 Troubleshooting IP telephones If the system is not properly configured several messages can appear Table 13 IP telephony display messages Message Description Solution SERVER NO PORTS LEFT The Business Communications Manager has run out of ports This message will remain on the 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 VoIP keycodes Se
118. configured to change when Daylight Savings Time occurs if the host Business Communications Manager is programmed to change Therefore if the telephone is in an area that does not change time for example Saskatchewan Canada you will need to readjust the time on your IP telephone at each time change You will also need to readjust the time if the IP telephone is in a time zone that changes and the Business Communications Manager is not for example if the telephone is in Alberta Canada and the Business Communications Manager is located in the business headquarters in Saskatchewan P0609327 02 Chapter 3 Installing IP telephones 71 Nortel Networks i2050 Software Phone The Nortel Networks 12050 Software Phone allows you to use a computer equipped with a sound card microphone and USB headset to function as an IP terminal on the Business Communications Manager system The Nortel Networks 12050 Software Phone uses the computer IP network connection to connect to the Business Communications Manager The registration process is the same as for the i200X telephones Registering the telephone to the system on page 48 When you install the Nortel Networks 12050 Software Phone on screen documentation walks you through the steps for installing the software You can also refer to the i2050 Software Phone Installation Guide Refer to the following section for details about configuring the Business Communications Manager connections for the N
119. cribing a fallback network on page 113 e Configuring routes for fallback on page 114 e Adding the destination code for the fallback route on page 116 e Example A private network configured for fallback on page 122 e Setting up the VoIP schedule to overflow on page 118 e PSTN fallback metrics on page 126 monitoring fallback calls By enabling PSTN fallback on the Local Gateway IP Interface screens for H 323 and SIP trunks you allow the system to check the availability of suitable bandwidth for a VoIP call then switch the call to a PSTN line if the VoIP trunk is not available or cannot produce the expected quality Refer to Setting up the local gateway on page 91 The Local Gateway IP Interface screen is accessed at Services IP Telephony IP Trunks H 323 Trunks or SIP Trunks You use scheduling and destination codes to allow the call to switch from H 323 and or SIP line pools to a PSTN line without requiring intervention by the user Use the dialing plan worksheet in the Programming Records to plan your dialing requirements so you can pinpoint any dialing issues before you start programming If you are programming an existing system you can look at what numbers the users are familiar with dialing and you can attempt to accommodate this familiarity into your destination codes plan The Programming Operations Guide provides configuration charts for various types of networks using PRI lines They can be adapted to VoIP
120. cription of restriction limitation 3 5 and prior Symbol portable IP handsets Login by Extension is login option offered by the telephone but is not currently supported by Business Communications manager The work around is to administer the extension as the username in Unified Manager The NetVision handsets do not support G 723 so they will be unable to negotiate a call on a VoIP trunk if the trunk is set to G 723 only Call Center ACD FEATURE 909 is not supported This is an unworkable feature on single line display sets including the M7100 and especially on Symbol Calls between Symbol sets do not support the Call Record feature There is sometimes significant echo heard on the Symbol set during ringback on outgoing calls over analog lines Business Communications Manager does not support remote registration for symbol sets if these sets are behind another device for example another Business Communications Manager or a third party router which has NAT turned on Each H323 Terminal configured utilizes one IP Client Resource whether the H323 Terminal is being used or not Firewall Default Rules when enabled block Symbol Registration and call processing You must add two additional rules 1 Pass Protocol TCP UDP Destination Port H 323 and 2 Pass Protocol UDP Destination port 1719 Ring cadence on Symbol handsets does not distinguish between Internal and External callers Symbol sets work fine as mem
121. curs the throughput decreases and the whole cycle repeats When multiple TCP sessions flow over few congestion links in the intranet the flow control algorithm can cause TCP sessions in the network to decrease at the same time causing a periodic and synchronized surge and ebb in traffic flows WAN links can appear to be overloaded at one time and then followed by a period of under utilization There are two results e bad performance of WAN links e IP telephony traffic streams are unfairly affected 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 na DiffServ network the Business Communications Manager system acts as a DiffServ edge device and performs packet classification prioritization and marking The router performs admission control for H 323 flows based on the WAN link bandwidth and utilization When received the WAN link marks the H 323 flows as Premium traffic and places the flows in the high priority queue Note Differentiated Service DiffServ is a QoS framework standardized by the Internet Engineering Task Force IETF e Inanon DiffServ or a legacy network the router manages the WAN link to make sure Premium VoIP packets have high priority in both directions when c
122. de VoIP service through Meridian 1 IPT BCM 3 6 and newer software or CSE 1000 gateways BCM 3 0 and newer software Destination Digits numeric Set the leading digits which callers can dial to route calls could be the through the remote gateway Ensure that there are no other same as the remote gateways currently using this combination of destination digits If multiple leading digits map to the same remote gateway separate them with a space For example 7 81 9555 These numbers are passed to the remote system as part of the dialed number 6 Click the Save button IP Telephony Configuration Guide 106 Chapter6 Setting up VoIP trunks for outgoing calls Configuring remote endpoints SIP trunks This section explains how to configure the Business Communications Manager to communicate with other Business Communications Managers VoIP gateways that accept the SIP trunk protocol version 3 5 software or newer Setting up the SIP address book Follow these steps to set up the SIP Address book for a remote gateway 1 In Unified Manager click the keys beside Services IP Telephony IP Trunks SIP Trunks 2 Click Address Book 3 On the top menu click on Configuration and select Add Entry Figure 31 Add an entry to the SIP address book Address Book Add Entry Neve Format 15 characters Delete Entry Destination IP 55e 255 255 255 QoS Monitor Disabled x Transmit Threshold o Receive Threshold o
123. ds to success By providing your customers with rapid access to sales and support personnel via telephone the Web and e mail your business can provide better customer service than ever before IP telephones and VoIP trunks This section describes two similar applications for IP telephony on the Business Communications Manager system IP telephones and VoIP trunks These applications can be used separately or together as a network voice data solution Refer to the information under the following headings e P telephones e VoIP trunks 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 through the Business Communications Manager can pass over VoIP trunks or across Public Switched Telephone Network PSTN lines Nortel Networks provides two types of IP telephones The IP telephones are wired to the IP network using Ethernet in the case of the i series 200X IP telephones or are accessed through your desktop or laptop computer as in the case of the Nortel Networks 12050 Software Phone Emobility voice can be provided using Symbol NetVision or NetVision Data telephones which connect through an access point wired to an IP network configured on the LAN NetVision telephones us
124. dwidth and average bandwidth requirements for a normal two way call must take into account the affects of full and half duplex links and the affects of silence suppression Refer to the tables in the next two sections below and to Table 40 on page 156 for voice Gateway bandwidth requirements Peak bandwidth is the amount of bandwidth that the link must provide for each call Considering voice traffic only the number of calls a link can support is Number of Calls Usable Link Bandwidth peak Bandwidth per call The average bandwidth takes into account the affects of silence suppression which over time tends to reduce bandwidth requirements to 50 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 167 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 Bandwidth requirements on half duplex links The following table provides bandwidth requirements for normal two way voice calls on a half duplex link for a variety of link pr
125. e 121 Figure 40 Example PSTN fallback 4 ucsdena Vou Ead ba sirrane da Ea ye 122 IP Telephony Configuration Guide 12 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 Figure 54 Figure 55 Figure 56 Figure 57 Figure 58 Figure 59 Figure 60 Figure 61 Figure 62 Figure 63 Figure 64 Fallback MEIGS OME i voor dede A323 deed eR EO aS 126 Fort ranges dialog DOE s2c04 aces Ween C ERPARERESEECITARPE PREEPTAED 130 FOR RANGES m P uH 131 xol c aa ist dale eke leg ae aah Maekawa eddie xk dedhdaa el deeded CT 131 Fort raies delbd DOK vecenontiricugreesacadesenidsdorsesers code 132 Business Communications Manager systems with a gatekeeper 141 IP trunking interoperability fields llle 143 NetMeeting GD OHS cccerctncert ei eee nee et bee kiateia shee named wae 145 NetMeeting Advanced Calling Options 0 0 0 0 eee ee 146 M1 to Business Communications Manager network diagram 148 Multiple Business Communications Manager systems network diagram 149 Routing all public calls through one Business Communications Manager 150 M1 to Business Communications Manager network diagram 151 Connecting to IP telephones 0 0 00 cece eee ens 152 LAN engineering peak transmission 00 0c e eee eee ees 157 Peak tame WAN IDK Sa zuo ERE RS dur dced
126. e Business Communications Manager can be connected to a LAN through a LAN card to a WAN through a WAN card and to a PSTN through trunk media bay modules as shown for Business Communications Manager A in the above diagram Through these networks the system accesses other systems and network equipment connected to the network P0609327 02 Chapter 1 Introduction 25 M1 IPT The Meridian 1 Internet Telephony Path M1 IPT allows Meridian 1 systems to communicate with the Business Communications Manager via H 323 trunks Telephones on the M1 such as Meridian telephone A can initiate and receive calls with the other telephones on the system across IP networks To provide fallback at times when IP traffic cannot pass you can also connect the Meridian to the Business Communications Managers through ISDN PRI SL 1 lines which provide the same MCDN capability that you can achieve through the H 323 VoIP trunks with MCDN active Refer to the Programming Operations Guide for a description of MCDN features and networking with PRI SL 1 lines Typical network applications using MCDN on page 147 describes how to provide the same network over VoIP lines A Business Communications Manager connected to an M1 IPT using the MCDN protocol can provide access to a central voice mail and call attendant systems which can streamline multi office telephony administration Telephones The Business Communications Manager can communicate using digital
127. e Pool Access dialog box appears 4 Type the letter of the VoIP line pool Click the Save button Repeat steps 4 and 5 if you have both H 323 and SIP line pools and you want to assign both to the telephone 7 Repeatthis procedure for every telephone you want to allow to use VoIP line pools If you plan to use fallback for your VoIP lines you need to configure the VoIP line pools into routes and assign a destination code for the route Refer to the Programming Operations Guide for details about creating routes and destination codes PSTN call to remote node Making a call to a remote node requires any Business Communications Manager systems between the calling and receiving nodes to have the correct routing to pass the call on to the next node For routing details on tandem networks refer to the Programming Operations Guide Private Networking section The following figure shows a call tandeming from the public network through System A Santa Clara and being passed to System B Ottawa In this case it might be a home based employee who wants to call someone in Ottawa You cannot program DISA or auto answer for VoIP trunks therefore your system cannot be accessed from an external location over a VoIP trunk The exception to this is if the call comes into a tandemned system system A from a PSTN and the call is then sent out across a VoIP trunk to system B as in this example In this case system A is controlling remote access thr
128. e QoS monitor data on page 128 e Viewing QoS monitoring logging on page 128 Quality of Service Status The QoS Status displays the current network quality described as a Mean Opinion Score MOS for each IP destination A pull down menu allows the administrator to view the MOS mapping The table below shows a sample QoS Monitor Table 28 QoS status IP G 711 G 711 G 723 1 5 3 G 723 1 6 3 aLaw uLaw kbit s kbit s G 729 QoS Monitor Tx Rx Tx Rx Tx Rx Tx Rx Tx Rx 47 192 5 2 Enabled 4 00 4 30 4 00 430 4 80 490 4 75 4 70 4 50 44 50 47 192 5 6 Disabled N A N A N A N A N A N A N A N A N A N A IP Telephony Configuration Guide 128 Chapter6 Setting up VoIP trunks for outgoing calls Note For the QoS monitor and PSTN fallback to function both Business Communications Managers must list each other as a Remote Gateway and QoS Monitor must be enabled on both systems Updating the QoS monitor data To update the table with the most current values From the View menu select Refresh Viewing QoS monitoring logging QoS monitor can be configured to log data The process for setting up logging is described in detail in the Programming Operations Guide The following steps explain how to view the log 1 2 3 On the Unified Manager navigation tree click the Services and Qos Monitor keys Click the Mean Opinion Score heading Click the Logging tab The Logg
129. e VoIP trunks are not available you must use routes and destination codes to access the VoIP trunk line pools Setting up remote gateways and end points This section explains how to set up your system to place calls through VoIP trunks The system at the other end of the call must be set up to receive VoIP calls For information about this refer to Outgoing call configuration on page 107 Programming for connecting Business Communications Managers together using PRI SL 1 lines and MCDN protocol is described in detail in the Programming Operations Guide Private Networking section VoIP trunks are configured in the same way with the addition of gateway programming required for IP trunks which is explained in the sections following Local gateway settings are described in Setting up the local gateway on page 91 For detail about outgoing call configuration view the information under the following headings e Configuring a remote gateway H 323 trunks e Configuring remote endpoints SIP trunks on page 106 P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 103 Configuring a remote gateway H 323 trunks 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 IPT using H 323 trunks The remote gateway list must contain an entry for every remote system to which you want to make
130. e a digit Santa Clara 3 oute 867 Vo Route 867 VoIP Route 774 PRI line Route 774 PRI line Route 009 PSTN p Public fallback line lt Route 009 PSTN line line with external Dialout with external Dialout 1613555 16135552244 1408555 Destination code 2 14085553322 Destination code 3 Route 867 absorb 0 Route 867 absorb 0 Route 774 absorb 0 Route 774 absorb 0 Destination code 9 Destination code 9 Route 009 absorb 1 Route 009 absorb 1 PSTN Business Communications Manager Santa Clara Business Communications Manager Ottawa Private IP address 10 10 5 1 Private IP address 10 10 4 1 e Public IP address 47 62 84 1 Public IP address 47 62 54 1 e DNs 3000 3999 e DNs 2000 2999 P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 123 From this system dial 9 to get onto PSTN e From this system dial 9 to get onto PSTN e Dialing plan CDP Dialing plan CDP destination code is part of DN Routing Routing e Target DN 2244 first digit is unique to Target DN 3322 first digit is unique to system system e Remote gateway destination digit 2 Remote gateway destination digit 3 Destination code 2 Destination code 3 e VolP private network dialout no external VolP private network dialout no external user dials 2244 no absorbed digits user dials 3322 no absorbed digits The sys
131. e an extended version of the H 323 protocol to connect to the system Note For this release NetVision telephones are not able to use SIP trunks 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 to a gateway on the remote system The device at the other end reassembles the packets into a voice signal Both H 323 and SIP trunks support private networking P0609327 02 Chapter 1 Introduction 23 between Business Communications Managers H 323 trunks can support connections to a number of different types of equipment including the Meridian 1 running IPT Succession 1000 M DMS100 switches and SL100 switches and trunk applications such as NetMeeting SIP trunks do not currently support the MCDN network protocol or interconnection with a Meridian system Creating the IP telephony network This section explains the components of the Business Communications Manager system and the devices it interoperates to create a network The information under the headings in this section describe the various components of the system e Networking with Business Communications Manager on page 24 e MI IPT on page 25 e Telephones on page 25 e Gatekeepers on the network on page 25 e IP network on page 26 e Public Switched Telephone Network on page 27
132. e firmware download Add Feature Modify Feature Delete Feature 6 A dialog box appears prompting you to proceed Click Yes to reset the password The password resets to Null The user can enter hot desking again to enter a new password IP Telephony Configuration Guide 62 Chapter3 Installing IP telephones Notes about Hot Desking The Hot Desking feature allows a user to divert calls and signals from one IP telephone to another For instance if a user is temporarily working in another office they can retain their telephone number by hot desking their usual telephone to the IP telephone in their temporary office e The headset mode is not transferred by this feature e Hot desking can be accessed using FEATURE 999 on the telephone to which the traffic will be diverted The user can also evoke this feature from the Services key menu where it is defaulted as the first item on the list Both telephones must be on hook before the feature can be used or cancelled e Hot desking must be allowed on the originating telephone and you need to specify a password These settings are found under the ADMIN key within the hot desking feature Hot desking is invoked through the DIVERT key within the hot desking feature e If the originating telephone does not have hot desking allowed the user will receive a Not allowed prompt indicating that the telephone is not available for hot desking This prompt also occurs if the originating telephone
133. e rr veoh eee b ex Y seer ORE ER PRS ec Oy ORA e det 19 Chapter 1 ipeo Mae 21 IP telephones and VOIP 1UNkS 2260 cc 66 seus ek ee 3 9 eee RR 22 IP Tels EON Sc Rohe REMI RARE Pee dpa cba qae fn a ONDE lof a2 VIP TUNKS oue desee eps E xu pk ver ea TTT 22 Creating the IP telephony network os dade seid eAWek xd RARE dd exa vd d d 23 Networking with Business Communications Manager 05 24 Ce een er ee er ee eee ee ee eee UM eres 25 TOISDHOUDS oodd qp Rid VOY ados Sw er gd av Edd qe E eX qd eq ecu 25 Gatekeepers on the network isssses sso hern n ERE ee 25 PROMO ouscexdrzondaa Kota ASTE SETRa dO EECENTAIPAESqUXISQI gages 26 Public Switched Telephone Network 0 0 c eee eee 27 Key IP telephony concepts ius cesta quk eee ke Soe RARE 27 enu o occ cerca SHR SESH SSPRI TIAA T ORME OE RMT ORME ERED EE 27 SAGE ERIT Fi sect hh ydo i ga ht hod A hy AE he ek ghia ke ih ied tim a 28 QOS Oll rreri vena k Seka saewtGess oebe dung RR EREIGAS RE ad dea 29 Chapter 2 Prerequisites checklist sssssacccae crn case awa esnene eae ene ne anaaas 31 ee SON co oes sani cdo hia eee anes aes au Network cp 2 32 Network ass DESI usa euemeerzirerbrer4 e e bepex ua pa eT er Rdas Ri 33 Resauice SSSGSIGN 4 idea tedideeeedeehoieees aoe eae meses bad E qe 33 ncn uo ci EDITI 34 System configuration Tor IP functions uuu cone bonnes RET XR BORSE x eR RO ROS RC 34 Finding the published IP address
134. e the Keycode Installation Guide Invalid Server Address The S1 is incorrectly configured with the IP address of a Business Communications Manager network adapter other than the published IP address IP Address conflict The telephone detected that a device on the network is currently using the IP address allocated to the telephone Registration Disabled The Registration on the Business Communications Manager is set to OFF SERVER UNREACHABLE RESTARTING Check that you have entered the correct Netmask and gateway IP addresses If the settings are correct contact your system administrator NEW SET The telephone has not been connected to the Business Communications Manager before and must be registered Programming note To see the configuration information for a telephone connected to the Business Communications Manager When the telephone is not on a call press the key bottom right corner of the telephone followed by the Ce key next to the key The display will automatically scroll through the configuration settings IP Telephony Configuration Guide 52 Chapter3_ Installing IP telephones To see the Codec data for a telephone while it is on a call Press the key followed by the Ca key Operation issues Here are a few possible issues you may encounter including a description of what may cause them and how to troubleshoot the issue Table 14 IP telephone troubles
135. e use the Nortel IP Terminals Configuration menu Deregistering a telephone using the IP record on page 66 e use the Configuration menu under one of the relevant headings under DN registration Deregistering a telephone using a DN registration heading on page 67 Deregistering a telephone using the IP record To deregister a DN for an IP telephone from the IP record 1 Inthe Unified Manager open Services IP Telephony IP Terminals and click Nortel IP Terminals The IP Terminal summary appears 2 Click the IP Terminal Status tab 3 Select the IP Terminal with the DN you want to deregister 4 Open the Configuration menu or alternate click anywhere on the listing for the terminal to display the menu as shown in the next figure P0609327 02 Chapter 3 Installing IP telephones 67 Figure 13 Deregister DN from Configuration menu Configuration Performance Fault Report Tools Logoff View Help Summary General IP Terminal Status Telephony Features List IP Terminal Status Offline Offline Offline Modify DUE Force firmware download Reset Hot Desking Password 5 Click Deregister DN 6 Reregister the telephone as described in Configuring the i20XX telephones to the system on page 47 A Warning After this feature is activated all active calls are dropped Deregistering a telephone using a DN registration heading To deregister a DN from a DN regi
136. eadset but the originating telephone does not once hot desking is activated the headset on the target telephone will no longer work To correct this situation you need to cancel hot desking plug a headset into the originating telephone then re establish hot desking at the target telephone Head set feature active on Originating telephone Target telephone target set after hot desking Yes l d No No l 0 Refer to the Telephony Features Handbook for details about using this feature Customizing feature labels When your IP telephone acquires a DN record the default settings are applied to the telephone including assigning features to the memory keys on the telephone These features all have pre defined labels and the telephone automatically displays the appropriate labels beside the programmed buttons If you want to customize these labels to be more appropriate you can do so through the Feature Labels heading on the Unified Manager The screens under the Feature Labels heading allow you to define custom labels for 24 features The system comes with 10 default labels which are feature and language specific depending to which region your system was assigned The default labels are mainly messaging and call attendant features However you can change any other feature label by adding to this list or by deleting any of the default settings and inserting new codes and labels Refer to Changing features or la
137. ectly directed to your system Refer to Using a gatekeeper on page 133 for details about configuring a gatekeeper to work with the Business Communications Manager e Outgoing call configuration on page 107 e Setting up VoIP trunks for fallback on page 112 e Quality of Service Monitor on page 127 The following figure shows a simple private networking configuration of three systems connected by VoIP trunks As in all private networking each system has direct routing configurations to the directly adjacent systems As well the dialing plans are configured to ensure that remote calls are correctly routed to the receiving system such as if Node A called someone in Node C IP Telephony Configuration Guide 102 Chapter6 Setting up VoIP trunks for outgoing calls Figure 29 Internal call from Meridian 1 tandems to remote PSTN line Business Communications Manager Calgary Node B VoIP trunk with T Ottawa o ERIS A MCDN S iia 3 A o een i nung js ag a o hi ee o Bggg s e A EC 5 VoIP trunk with MCDN I C Noda A Meridian 1 Headoffice Since the VoIP trunks are configured into line pools you can assign line pool codes to users who have been assigned access to the VoIP trunks However if you intend to set up your system to use fallback so that calls can go out over land lines if th
138. ed GateKeeperResolved or GateKeeperRoutedNoRAS are selected under Call Signaling type the IP address of the machine that is running the gatekeeper Backup gatekeeper s IP address gt NetCentrex gatekeeper does not support RAS IP address therefore any backup gatekeepers must be entered in this field Note Gatekeepers that use RAS can provide a list of backup gatekeepers for the end point to use in the event of the primary gatekeeper failure Alias Names If GateKeeperRouted GateKeeperResolved or GatekeeperRoutedNoRAS are selected under Call Signaling type one or more alias names for the gateway One or more alias names may be configured for a Business Communications Manager Alias names are comma delimited and may be one of the following types E 164 numeric identifier containing a digit in the range 0 9 Identified by the keyword TEL Example the Business Communications Manager is assigned an E 164 and an H323 Identifier A1ias Names TEL 76 NAME bcm10 nortel com e NPI TON also referred to as a PartyNumber alias Similar to E164 except that the keyword indicates the NPI numbering plan identification as well as the TON type of number Identified by one of the following keywords PUB Public Unknown Number PRI Private Unknown Number UDP Private Level 1 Regional Number UDP cDP Private Local Number CDP Refer to Notes about NPI TON aliases for H 323 trunks on page 96 e H
139. ee ee eee 71 Configuring the i2050 Software Phone for the local system 71 Chapter 4 Installing NetVision telephones 0 00 cee eee eee eee 73 Nel Vision CORITBOIVI ss opor see OR RELTISSIICAPLAGZCEXR PARREVESDILAR CHER 4 73 P0609327 02 Contents 5 PODES DOMME dde qaxaxd wa qpePe ada haa heotaeeeeeaeeee eee yee es 73 CEE Ce a ee ee eee ee E E ee ee 74 Handsetand call TIEDUODBS cat zemRx oeede Hoes HOLS ORE 9S4 OREEE HRS S 74 sco eT TT eee T 74 Configuring NebvIsion records 205546 sco ker RR e iredi karua RA OR RC hn 75 Gathering system information before you start us anaana aaan 75 Assigning general settings isseccasexkkestirrckrer RR 4d RPG Geek 43s 76 Monitoring H 323 service Stals ucesoqudqaeaiqe e PI Eni XEPEP ATE x mes T77 Assigning H 323 Terminals records 0 00 cece eee 78 Pre configuration notes for NetVision handsets 2 78 Adding a NetVision record in the Unified Manager 79 Modifying H 323 terminal records 34s lt ees sedis re e dads beens RE ons 81 Updating the H 323 terminals record ciues ossa suae sacco ak m RE xo 81 Changing a handset Name 24 0200 c080cc0es ee e yh ey 82 Changing the DN record of ahandset 0 000 cee eee eee 82 Deleting a NetVision telephone from the system llle lessen 82 Deregistenng a telephone uus c xe xem Rer RC ons e RACER RC n ACA d 83 Chapter 5 Configuring local VoIP trunks
140. egabits per second Millions of bits per second MCDN This is a specific network protocol used on private networks between Business Communications Manager systems or between Business Communications Manager systems and Meridian systems The protocol only works on PRI SL 1 lines and on VoIP trunks The protocol is activated with a keycode modem 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 IP Telephony Configuration Guide 208 Glossary network diagram This is a physical drawing description of how the local network works to which your Business Communications Manager will be connected It also includes information about the Business Communications Manager requirements such as public and or private IP addressing DHCP requirements and quality of service availabilities Where possible it should include information about the public networks and any changes or adjustments required by the network or the Business Communications Manager for compatibility Nortel NetVision Phone Administrator NVPA This is the Business Communications Manager specific application that is used to configure features and system information into the NetVision handsets This application is included on the Business Communications Manager database The latest application can be obtained at http www symbol com services dow
141. elephone 1 Inthe Unified Manager open Services IP Telephony IP Terminals and click Nortel IP Terminals The IP Terminal summary appears 2 Click the IP Terminal Status tab 3 Select the listing for the IP telephone you want to upgrade 4 Open the Configuration menu or alternate click anywhere on the listing for the terminal to display the menu 5 Select Force Firmware Download A message appears that asks you want to confirm that you want to proceed 6 Click the Yes button The firmware download begins IP Telephony Configuration Guide 66 Chapter3_ Installing 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 complete and the telephones have reset Note In order not to saturate the IP network with download packets the system will only download up to five IP telephones at any given time Telephones requiring download will show a Unified Manager status of Download Pending and the UNISTIM Terminal Proxy Server UTPS will initiate download as resources become available Deregistering DNs for IP telephones You can deregister selected telephones from the Business Communications Manager and force the telephone to go through the registration process again i Warning After this feature is activated all active calls are dropped There are two ways to deregister an IP telephone
142. elephone G 711 with VAD that is on a remote or busy sub net Refer to Choosing a G 729 codec on page 45 G 729 with VAD Note You can only change the codec on a configured IP G 723 telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone JitterBuffer Auto Increase the jitter buffer size for any telephone that has poor Default network connectivity to the Business Communications None Manager en Refer to Choosing a Jitter Buffer on page 46 jos Note You can only change the jitter buffer on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone 6 Click the Save button P0609327 02 Chapter 3 Installing IP telephones 59 Working with the features list You can add and modify the features that display on the IP telephone feature list which is accessed through the Services button or by using FEATURE 900 Refer to Using the Services button to access features on page 60 The Programming Operations Guide provides a complete list of Business Communications Manager Features and index codes The Telephony Features Handbook provides details about using the features Note that the list assigns the hot desking feature to position 1 refer to Resetting the Hot Desking password on page 61 1 Inthe Unified Manager open Services IP Telephony and click Nortel I
143. elephone Refer to the installation card that comes with the telephone for specific connection directions Plug the AC Power adapter into the base of the telephone and then plug the adapter into the AC outlet Once the telephone is connected refer to Configuring the 120XX telephones to the system Configuring the i20XX telephones to the system Configuring Nortel IP telephones involves two processes If DHCP Distributed Host Control Protocol service on the Business Communications Manager is active or the Customer DHCP server has been configured to hand out the specific Business Communications Manager details the IP telephone will automatically attempt to find the server Refer to Configuring DHCP on page 53 which describes the specific DHCP requirements for IP telephones and to the Programming Operations Guide which provides detailed DHCP configuration information After you register the telephone to the system as described in Registering the telephone to the system the telephone assumes the parameters it receives from the system which are described in Configuring telephone settings If DHCP is not configured to provide system information or if you are not using DHCP on your network you need to configure your telephone parameters before the telephone can register to the system In this case follow the directions in Configuring telephone settings and then follow any of the prompts that appear as described in Reg
144. elephony Prioritization This section includes information about setting up earlier versions of BayStack routers and how to set up a range of UDP as a high priority Note The information in this section is not required for recent versions of the Nortel Networks routers such as BayRS release 15 that support prioritization based on the DiffServ Code Point DSCP Creating an outbound traffic filter To create an outbound traffic filter 1 In the Configuration Manager window a Click Circuits and select Edit Circuits The Circuit List window appears b Selecta circuit C Click the Edit button The Circuit Definition screen appears with the circuit you selected highlighted On the Protocol menu a Click Add b Select the protocol priority from the list C Click the OK button Click Protocols Edit Protocol Priority and then click Priority Outbound Filters The Priority Outbound Filters window appears Click Template The Filter Template Management window appears a Enter the template name and click Create The Create Priority Outbound Template window appears b Type a descriptive name in the Filter Name field Click Criteria Add Datalink IP and then click Criterion The Add Range window appears If you choose the User Defined criterion the Add User Defined Field window appears first Type a minimum and maximum value to specify the range and then click the OK button The Add Range window closes The new criterion and ran
145. em reboot and can only be manually deactivated Deactivating the VoIP schedule Follow these steps to deactivate a schedule 1 Dial FEATURE 873 The phone prompts you for a password 2 Type the password 3 Press OK The system returns to the Normal schedule IP Telephony Configuration Guide 120 Chapter6 Setting up VoIP trunks for outgoing calls How fallback routing works CDP network User dials 82233 remote system DN 2233 remote identifier destination digit 2 The system absorbs the 8 and dials out 2233 If the call falls back to PSTN line the system still only absorbs the 8 If the PSTN line is on a private network the system dials out 2233 If the PSTN line is a public line the system dials out the public access number to the remote system in front of the 2233 Refer to Figure 38 and Figure JB Figure 38 Setting up routes and fallback for call to remote system CDP dialing code Note For this example the destination code to call inside the other system is 82 Both systems have destination code 9 set up as the local PSTN access code Call to person within remote BCM system User dials DN which includes the destination code Example 82233 Create route to VoIP pool Create route to PSTN pool Route XXA VoIP pool Route XXB PSTN line pool No external External lt public to remote DN type Private DN type Public Destination code 82 schedules Assign VO
146. en H 323 and SIP trunks 7 d Have you set up line parameters determined line pools for H 323 and or SIP trunks and set up routing and destination codes Have you determined which system telephones will have access to these routes 7 e If you have not already assigned target lines have you defined how you are going to distribute them on your system 7 f Have you decided if you are going to employ the fallback feature If yes ensure that your routing and scheduling are set up Ensure that QoS is activated Network note If your Business Communications Manager is part of a private network have the other Business Communications Managers in the network been upgraded to BCM 3 5 or newer software or had QoS patch 3 0 0 25 or later applied If there is a Meridian 1 on the network is it running IPT 3 0 or newer If either of these conditions are not met your H 323 trunks will not work correctly Refer to Configuring local VoIP trunks on page 85 Setting up VoIP trunks for outgoing calls on page 101 and Optional VoIP trunk configurations on page 129 for detailed configurations IP Telephony Configuration Guide 40 Chapter2 Prerequisites checklist IP telephone records Complete this section if you are installing i series and or NetVision telephones Table 10 IP telephone provisioning Prerequisites Yes No 8 a Are IP connections and IP addresses available for all IP telephones If yo
147. enabled the half duplex nature of normal voice calls allows the sender and receiver to share the same bandwidth on the common channel While the sender is talking the receiver is quiet Since only one party is transmitting at a time silence suppression reduces the peak bandwidth requirement per call on a half duplex link to Peak Bandwidth per call 1 Continuous Transmission Rate Half Duplex links With Silence Suppression Bandwidth requirements on full duplex links The following table provides bandwidth requirements for normal two way voice calls on a full duplex link for a variety of link protocols codec types and payload sizes Bandwidths for full duplex links are stated in terms of the individual transmit and receive channels For instance a 64 kbits full duplex link e g a DSO on T1 link has 64 kbits in the transmit direction and 64 kbits in the receive direction Table 40 Bandwidth Requirements per Gateway port for Full duplex links Ethernet B W PPP B W FR B W Payloa NoSP Silence NoSP Silence NoSP Silence d Size Suppression Suppression Suppression ms peak peak Avg peak peak Avg peak peak Avg Codec Type kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s G 711 30 81 6 81 6 81 6 75 2 75 2 75 23 77 1 77 1 77 13 64 kb s G 729 30 25 6 25 6 12 8 19 2 19 2 9 6 21 1 21 1 10 6 8 kb s G 723 1 30 24 0 24 0 12 0 17 6 17 6 8 8 19 5 19 5 9 8 6 3 kb s
148. end points on page 102 When prompted for the IP address of the remote gateway type the IP address of the client computer Repeat this procedure for every NetMeeting client you want to set up P0609327 02 147 Chapter 8 Typical network applications using MCDN This section explains several common installation scenarios and provides examples about how to use VoIP trunks and IP telephony to enhance your network Information in this section includes e Setting up MCDN over VoIP with fallback on page 147 e Networking multiple Business Communications Managers on page 149 e Multi location chain with call center on page 151 Business Communications Manager to remote IP telephones on page 152 Setting up MCDN over VoIP with fallback The MCDN networking protocol between a Meridian 1 and one or more Business Communications Managers works the same way as it does over PRI lines You still require the MCDN and IP telephony software keys and compatible dialing plans on all networked systems The one difference between MCDN over PRI and MCDN over VolP is that the VoIP trunks require specific Remote Gateway settings unless there is a Gatekeeper configured to route traffic on the IP network Under Services IP Telephony H 323 Trunks Remote Gateway ensure that Gateway Protocol is set to CSE for the VoIP connection to the Meridian 1 IPT system The Gateway Type would be set to IPT as it would for any non MCDN VoIP co
149. ending on CODEC type and number of frames per packet to introduce a 60 millisecond delay e MEDIUM 120 millisecond delay e LARGE 180 millisecond delay For information about choosing a Jitter Buffer refer to Choosing a Jitter Buffer on page 46 P0609327 02 Chapter 3 Installing IP telephones 45 Table 11 IP terminals general record fields Continued Field ms Value Description G 729 Payload Size 10 20 30 40 50 Setthe maximum required payload size per codec for the IP 60 telephone calls sent over H 323 trunks Default 30 Note Payload size can also be set for Nortel IP trunks Refer ms G 723 Payload Size 30 to Configuring media parameters on page 89 ms G 711 Payload Size 10 20 30 40 50 60 Default 20 4 Goto Installing i series telephones on page 46 Choosing a codec The default codec is used when an IP client has not been configured to use a preferred Codec Refer to the next section for individual IP client Codec settings If the default Codec is set to AUTO the Business Communications Manager will choose the appropriate CODEC when an IP client makes a call For example if both endpoints of the call are IP telephones on the same subnet the Business Communications Manager chooses G 711 for maximum voice quality If the telephones are on different subnets the Business Communications Manager will choose G 729 to minimize network
150. ends on the voice packet size and the link bandwidth and is the result of the following formula serialization delay in ms 8 IP packet size in bytes link bandwidth in kbit s Queuing delay The queuing delay is the time it takes for a packet to wait in the transmission queue of the link before it is serialized On a link where packets are processed in a first come first served order the average queuing time is in milliseconds and is the result of the following formula queuing time in ms 8 average IP packet size in bytes 1 p link bandwidth in kbit s The average size of intranet packets carried over WAN links generally is between 250 and 500 bytes Queueing delays can be important for links with bandwidth under 512 kbit s while with higher speed links they can allow higher utilization levels Routing and hop count Each site pair takes different routes over the intranet The route taken determines the number and type of delay components that add to end to end delay Sound routing in the network depends on correct network design Reduce link delay In this and the next few sections the guidelines examine different ways of reducing one way delay and packet loss in the network The time taken for a voice packet to queue on the transmission buffer of a link until it is received at the next hop router is referred to as the link delay Methods to reduce link delays include Upgrade link capacity to reduce the serialization delay of the
151. equired as part of the network to which your Business Communications Manager system is attached but Gatekeepers can be useful on networks with a large number of devices Referring to Figure on page 24 for example Digital telephone A wants to call IP telephone B which is attached to Business Communications Manager B over a network that is under the control of a gatekeeper IP Telephony Configuration Guide 26 Chapter1 Introduction Digital telephone A sends a request to the gatekeeper The gatekeeper depending on how it is programmed provides Digital telephone A with the information it needs to contact BCM B over the network Business Communications Manager B then passes the call to IP telephone B SIP trunks do not use gatekeepers The Business Communications Manager does not contain a gatekeeper application If you want to put a gatekeeper on your network it must be put on a separate gatekeeper server The Business Communications Manager is compatible with RadVision CSE 1000 CSE1K and NetCentrex gatekeepers Refer to Using a gatekeeper on page 133 and Appendix D Interoperability on page 175 i Warning Meridian 1 IPT does not support the RadVision gatekeeper IP network In the network shown in Figure 1 on page 24 several LANs and a WAN are shown When planning your network be sure to consider all requirements for a data network Your network administrator should be able to advise you about the network
152. er about configuring DNs in the Programming Operations Guide for details For any telephones assigned with VoIP line pools that do not have target lines assigned go to Incoming calls Assigning target lines on page 98 and configure target lines for these telephones To configure incoming traffic refer to Configuring media parameters on page 89 To configure outgoing traffic refer to Chapter 6 Setting up VoIP trunks for outgoing calls on page 101 P0609327 02 Chapter 5 Configuring local VoIP trunks 89 Configuring media parameters You can use the screen described in this section to determine the order the VoIP trunk will select codecs the silence suppression settings and the jitter buffers 1 In Unified Manager click the Services IP Telephony IP Trunks keys 2 Click H 323 Trunks or SIP Trunks depending on the type of trunk you want to configure 3 Click the Media Parameters tab The Media Parameters dialog appears Figure 23 H 323 Media Parameters dialog box 9 IP Telephony System Configuration IP Terminals 9 IP Trunks gt o SIP Trunks Media Parameters Media Parameters 1st Preferred Codec a723 v 2nd Preferred Codec a73 3rd Preferred Codec G7ituaw 4th Preferred Codec G7itaaw gt Silence Compression Disabled gt Jitter Buffer Voice aro OOO T 38 Fax Support Enabled G 729 Payload Size ms o vj G 723 Payload Size ms
153. er 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 Further network analysis This section describes how to examine the sources of delay and error in the intranet It also discusses several methods for reducing one way delay and packet loss The key methods are described under the following headings e Components of delay on page 163 e Reduce link delay on page 163 e Reducing hop count on page 164 e Routing issues on page 165 P0609327 02 Efficient Networking 163 Components of delay End to end delay is the result of many delay components The major components of delay are Propagation delay Propagation delay is the result of the distance and the medium of links moved across Within a country the one way propagation delay over terrestrial lines is under 18 ms Within the U S the propagation delay from coast to coast is under 40 ms To estimate the propagation delay of long haul and trans oceanic circuits use the rule of thumb of 1 ms per 100 terrestrial miles If a circuit goes through a satellite system estimate each hop between earth stations adds 260 ms to the propagation delay 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 dep
154. eriods of silence during speaker pauses between words and phrases By applying silence compression average bandwidth use is reduced by the same amount This 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 effects of silence compression on peak bandwidth requirements differ depending on whether the link is half duplex or full duplex IP Telephony Configuration Guide 168 Silence compression Silence compression on half duplex links The following figure shows the bandwidth requirement for one call on a half duplex link without silence compression Since the sender and receiver share the same channel the peak bandwidth is double the full transmission rate Because voice packets are transmitted even when a speaker is silent the average bandwidth used is equal to the full transmission rate Figure 58 One call on a half duplex link without silence compression Tx Hello Fred This is Susan Do you have a minute R Fred here Hi Sure x Conversation P4 Bandwidth used 2 ER mm SS mee SS Voice frames sent even when speaker is silent When silence compression is enabled voice packets are only sent when a speaker is talking In a typical voice conversation while one speaker is talking the other speaker is listeni
155. erminal Status LIN al E P Addre ode W Version itterBuffe e 2431 Offline i2050 N A Default N Default N A 2432 Offline i2004 N A Default N A Default N A 2433 Offline i2002 N Default N A Default N 3 Select the IP Terminal record for which you want to change the properties 4 Open the Configuration menu or alternate click anywhere on the terminal listing to open the Configuration menu and select Modify parameters nfic Tr Performance Faull Deregister DN Force firmware download Reset Hot Desking Password Add Feature Modify Feature Delete Feature The IP Terminal Status dialog box appears as shown in the figure below IP Telephony Configuration Guide 58 Chapter 3 Installing IP telephones Figure 8 IP Terminal status dialog box IP Terminal Status DN Status Type IP Address Codec F w Version JitterBuffer Terminal ID 2431 Read Only Field Offline i2050 N Default N Default N 5 You can change the Codec or JitterBuffer settings for the terminal All other fields are read only The table below describes the two configurable fields on this screen Table 16 IP Terminal Status fields Field Value Description Codec Default Specifying a non default CODEC for a telephone allows you G 711 aLaw to override the general setting You might for example want G 711 uLaw to specify a low bandwidth CODEC 9 729 for a t
156. ers on the public network to find the Business Communications Manager For example if a Business Communications Manager has a LAN interface LAN1 that is connected only to local office IP terminals and a WAN interface WANI that is connected to the public network then WANI should be set to the published IP address Setting the Global IP published IP To set the published IP address 1 In Unified Manager click on the keys beside Services and IP Telephony 2 Click IP Terminals The Global IP Setting tab appears as shown in the diagram below Figure 2 Global IP settings Global IP Setting 9 IP Telephony System Configuration Q IP Terminals Published IP Address IP LAN1 Global IP Setting Published IP Address 3 From the Published IP Address menu select the appropriate network interface Table 7 Published IP Address options Option Description IP LAN1 Choose the LAN number that corresponds with the LAN IP LAN 2 card you are using for this network IP WAN1 Choose the WAN number that corresponds with the WAN IP WAN2 card you are using for this network IP UTWAN1 If you are using a WAN connection using a Universal T1 line choose this option IP Telephony Configuration Guide 36 Chapter2 Prerequisites checklist Determining the published IP address Use the flowchart in the following figure to determine which card should be set as the published IP addre
157. erts you to important information B Tip Note Tip symbol Alerts you to additional information that can help you perform a task Security Note This symbol indicates a point of system security where a default should o be changed or where the administrator needs to make a decision about the level of security required for the system Text conventions This guide uses these following text conventions angle brackets lt gt bold Courier text italic text bold text plain Courier text Represent the text you enter based on the description inside the brackets Do not type the brackets when entering the command Example If the command syntax is ping ip address youenter ping 192 32 10 12 Represent command names options and text that you need to enter Example Use the dinfo command Example Enter show ip alerts routes Represents terms book titles and variables in command syntax descriptions If a variable is two or more words the words are connected by an underscore Example The command syntax show at valid route valid route is one variable and you substitute one value for it Represents fields names field entries and screen names in the Unified Manager application Represents command syntax and system output such as prompts and system messages Example Set Trap Monitor Filters P0609327 02 Preface 17 Acronyms This guide uses the following acronyms API AT
158. ervices the VoIP line pool Dialout N A Call routing Routes Route XXX Create a destination code that matches the Destination Digit s Destination code Destination code 2 3 Services Telephony Services Call routing Destination codes Define the Normal and VoIP shedules Normal Route 774 Absorb 0 digits VoIP Route 867 Absorb 0 digits Services Telephony Services Call routing Destination codes X Schedules Confirm or create a destination code for the PSTN Define Normal and VoIP schedules Destination code 9 Normal Route 009 absorb All digits VoIP Route 009 absorb All digits Services Telephony Services Call routing Destination codes 9 Schedules Activate the VoIP schedule from the control set 3321 2221 FEATURE 873 Making calls through a private VoIP network gateway From a telephone on Business Communications Manager Ottawa a caller dialing to a telephone on Business Communications Manager Santa Clara must dial the destination code which includes the destination digits for the Business Communications Manager Santa Clara remote gateway and the DN of the telephone For example dialing 3322 would connect as follows e 3isthe destination code If a suitable level of QoS is available the call is routed through the VoIP trunks and through the remote gateway with a destination digit of 3 The call is sent across the PDN using the IP address of the
159. es Radvision 135 assessment network 33 resources prerequisite 33 asymmetrical media channel negotiation 178 routing 165 Asynchronous Transfer Mode ATM 154 background noise 172 bandwidth available for other data 172 characteristics 154 determining requirements 153 full duplex links 156 half duplex link silence suppression 156 half duplex links 155 peak 155 silence compression 167 spare bandwidth 154 before you start IP telephony and network prerequisites 31 NetVision 75 block IP telephone dialout 52 bottlenecks 162 bridges network prerequisites 31 buffer jitter 46 buffers VoIP trunks 91 Business Communications Manager call chain network configuration 151 connecting to remote IP telephones 152 gateway router support 199 H 323 gateway specifications 176 MCDN system requirements 148 network device prerequisites 32 networking multi locations with call center 151 networking multiple systems 149 port settings 129 signaling method 91 system configuration prerequisites 34 using a gatekeeper 133 using firewalls 129 busy tone VoIP gateway progress tones 178 C call center networking multi locations 151 call chain network configuration 151 call progress tones 178 Call Signaling Local Gateway 93 call signaling modifying 92 calls gatekeeper examples 141 incoming configuration 98 making 125 media path redirection 177 capacity engineering link capacity 161 insufficient 162 Caution symbol 15 CDP network di
160. es each of which use a Business Communications Manager A typical use of this would be a 1 800 number that users world wide can call who are then directed to the remote office best able to handle their needs Figure 53 M1 to Business Communications Manager network diagram Call Center PSTN fallback route Intranet VoIP trunk Branch Offices i2004 telephone To set up this system 1 Ensure that the existing network can support the additional VoIP traffic 2 Coordinate a Private dialing plan between the systems 3 On each Business Communications Manager 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 Reboot each system 5 Set up a Call Center on the central Business Communications Manager IP Telephony Configuration Guide 152 Chapter8 Typical network applications using MCDN Business Communications Manager to remote IP telephones You can also set up a system that allows home based users or Call Center agents to use the full capabilities of the Business Communications Manager including access to system users applications and PSTN connections This system does not require VoIP trunk configuration This system functions in a similar manner to the system described in Multi location chain wit
161. es which call signaling method to use Refer to Using a gatekeeper on page 133 IP Telephony Configuration Guide 92 Chapter5 Configuring local VoIP trunks SIP trunks communicate between Business Communications Managers The addressing for the remote destination is described in Setting up SIP trunk subdomain names on page 97 Modifying local gateway settings for H 323 and SIP trunks To modify the settings for your local gateway 1 Inthe Unified Manager click the keys beside Services IP Telephony IP Trunking 2 Click H 323 Trunks or SIP Trunks depending on what type of VoIP trunk you are configuring The Local Gateway IP Interface screen for that type of trunk appears if you selected H 323 trunks If you selected SIP trunks click on the Local Gateway IP Interface tab Figure 25 Local gateway IP interface H 323 Trunks 9 IP Telephony Local Gateway IP Interface Q System Configuration Local Gateway IP Interface IP Terminals IP Trunks Fallback to Circuit Switched Enabled All 9 Remote Gateway Call Signaling Direct v SIP Trunks PortRanges Primary Gatekeeper IP noon Backup Gatekeeper s ooo R iCSSYN CONN Alias Names rone Registration TTL Seconds on Gateway Protocol CSE H245 Tunneling Disabled Call Signaling Port 1755 RAS Port o Figure 26 Local gateway IP interface SIP trunks I
162. estination codes 3 Enter one or more digits for this destination code Note For example if it is available you might want to use the same number that you used for the destination code of the gateway If you have multiple gateways you could use a unique first number followed by the destination digits to provide some consistency such as 82 83 84 85 to reach gateways with destinations digits of 2 3 4 and 5 P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 117 The number you choose will also depend on the type of dialing plan the network is using Networks with CDP dialing plans have unique system codes However with networks using UDP this is not always the case therefore you need to be careful with the routing to ensure that the codes you choose are unique to the route This will also affect the number of digits that have to be added or absorbed It is helpful to use the Programming Records to plan network routing so you can determine if there will be any conflicts with the destination codes you want to use 4 Click the Save button to close the dialog box Configuring the schedules for the destination codes Under the destination code heading you created in the previous section click the Schedules key then choose the appropriate schedules VoIP schedule for all fallback destination codes Figure 36 VoIP schedule 5A VoIP First Route Route 500 Absorbed length Second Route
163. eteteeu bkes 4 e465 dG Red aahed 158 Calculating network load with IP telephony traffic 04 160 Network loading bandwidth ananuna anaana 161 One call on a half duplex link without silence compression 168 One call on a half duplex link with silence compression 168 Two calls on a half duplex link with silence compression 169 One call on a full duplex link without silence compression 170 One call on a full duplex link with silence compression 171 Two calls on a full duplex link with silence compression 172 Relationship between users and services nsanra 192 P0609327 02 13 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 Table 22 Table 23 Table 24 Table 25 Table 26 Table 27 Table 28 Table 29 Table 30 Table 31 Table 32 Table 33 Table 34 Table 35 Table 36 Table 37 Table 38 Table 38 Table 39 Network diagram prerequisites lees 31 Network device checklist uasa seas eur dox i ica Ct xou Red Rc E Re deed Reed 32 Network assesment cccumauaeecm EX XU XS ER GHEE eee REFERS Oe DA 33 Resource dssb SIBI ivssduosipsiesbewesxeswurdwrad deaur CR d Arg 33 Keene ccssaxiededesuaieriseeexk Mp qe9292343 ped dag dur
164. etween two networks using the same addressing structure A bridge filters out packets that remain on one LAN and forwards packets for other LANs codec Equipment or circuits that digitally code and decode voice signals Software that provides compression decompression algorithms for voice traffic over IP networks and VoIP trunks For IP telephones the Business Communications Manager supports both a law and mu law variants of the G 711 CODEC as well as the G 729 and G 723 CODECS 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 Choose the G 711 CODEC with the companding law alaw or ulaw that matches your system requirements 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 The G 723 CODEC should be used only with third party devices that do not support G 729 or G 711 Codecs with VAD Voice Activity Detection make VAD active on the system which performs the same function as having silence suppression active Note You can only change the codec on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone IP Telephony Configuration Guide 204 Glossary communications protocol A set of agreed upon communications formats and procedures between devices on a
165. f restrictions and requirements refer to SIP trunk interoperability issues on page 188 H 323 network applications considerations In order to maintain a level of quality of transmission over VoIP trunks QoS monitor must be enabled and configured Refer to Configuring a remote gateway H 323 trunks on page 103 and Quality of Service Monitor on page 127 If your network uses a gatekeeper H 323 trunks only there are also specific settings that must be set on the Local Gateway screen to recognize the gatekeeper and also within the gatekeeper application so that VoIP lines are recognized Refer to Using a gatekeeper on page 133 If there is a gatekeeper on the network you do not have to configure remote gateway settings P0609327 02 Chapter 5 Configuring local VoIP trunks 87 If you plan to use H 323 trunking and you have a firewall set up ensure that the ports you intend to use have been allowed Refer to Incoming calls Assigning target lines on page 98 Chapter 8 Typical network applications using MCDN on page 147 provides examples of VoIP trunks used in private networking Warning Ensure that all systems in your network are either running BCM 3 5 or newer A software or have the QoS patch installed that allows them to interoperate with BCM 3 5 or newer software Systems running BCM software previous to 3 5 which do not have this patch installed cannot support VoIP trunks with systems ru
166. ft blank no password prompt occurs during registration Auto Assign DN On If set to ON the system assigns an available DN as an IP Off terminal requests registration It does not prompt the installer to enter a set DN Note For this feature to work Registration must be ON and Password must be blank If set to OFF the installer receives a prompt to enter the assigned DN during the programming session Note Refer to the Caution notice at the top of this section Advertisement Logo lt alphanumeric Any information in this field appears on the display of all IP string gt telephones For example your company name or slogan Default Codec Auto If the IP telephone has not been configured with a preferred G 711 aLaw codec choose a specific codec that the IP telephone will use when it connects to the system G 711 uLaw G 729 If you choose Auto the IP telephone selects the codec ras For information about choosing a codec refer to Choosing a codec on page 45 G 729 VAD If you are unsure about applying a specific codec ask your G 723 VAD network administrator for guidance Default Jitter Buffer None Choose one of these settings to change the default jitter Auto buffer size Small e NONE Minimal latency best for short haul networks Medium with good bandwidth L e AUTO Business Communications Manager will arge dynamically adjust the size e SMALL Business Communications Manager will adjust the buffer size dep
167. g IP Telephony Configuration Guide 38 Chapter2 Prerequisites checklist Table 8 IP terminals general record fields Continued Field Value Description G 723 1 Data Rate 5 3 kbps 6 3 kbps Choose the preferred data rate for the channel G 723 1 Data Rate 6 3 kbps 5 3 kbps rved Media Gateway Codec G 723 1 Data Rate selects what data rate is used for transmissions from the Business Communications Manager to an IP device when the G 723 1 family codec is used G 723 1 or G 723 1A This has no effect on any other codec The possible values are 5 3 kbps and 6 3 kbps T 38 UDP Redundancy 0 1 2 3 If T 38 fax is enabled on the system this setting defines how many times the message is resent during a transmission in order to avoid errors caused by lost T 38 messages Default 0 Reserved Media Gateway Codec G 711 G 729 G 723 Choose the preferred codec that you are using with your IP network Reserved Media Gateway Codec G 711 v 3 711 G 723 G 723 Reserved Media Gateway Codec should be set to whatever is the most commonly used codec for Media Gateways It determines the amount of codec resources reserved for each Media Gateway Reserving resources speeds up establishment of connections For example if most calls through a Media Gateway use the G 711 codec set this to G 711 If most calls use G 729 set this to G 729 Note that the higher the setting G
168. generates a summary of packet loss with a granularity of 2 and for each successful probe that made its round trip that many rtt samples For a strong network there must be at least 3000 delay samples and 60 packet loss samples Store the raw output of the Ping results in a file Determine the average and standard deviation of one way delay and packet loss Repeat this for each site pair At the end of the measurements the results are as shown in the following table Table 53 Delay and error statistics Measured one way delay ms Measured packet loss Expected QoS level Destination pair Mean Mean o Mean Mean o Mean Mean o Santa Clara 171 179 2 2 3 Good Good Richardson Santa Clara Ottawa Santa Clara Tokyo Richardson Ottawa Richardson Tokyo Ottawa Tokyo Other measurement considerations The Ping statistics described above measure the intranet before IP telephony installation The measurement does not take into consideration the expected load provided by the IP telephony users If the intranet capacity is tight and the IP telephony traffic is important the installer or administrator must consider making intranet measurements under load Apply load using traffic generator tools The amount of load must match the IP telephony offered traffic estimated in the Business Communications Manager VoIP Gateway Bandwidth requirements P0609327 02 Quality of Service
169. ges now appear in the Filter Information field of the Create Priority Outbound Template window Click Action Add and then click action IP Telephony Configuration Guide 180 Interoperability 8 Click the OK button The Filter Template Management window opens The new template appears in the templates list 9 Click Done The Priority Outbound Filters window opens 10 Click Create The Create Filter window opens a b c d Select a circuit in the Interfaces field Select a template in the Templates field Type a descriptive name in the Filter Name field Click the OK button The Priority Outbound Filters window opens 11 Click the Apply button The filter is applied to the circuit Sample criteria ranges and actions for UDP filtering The filtering goal is to place all VoIP H 323 traffic leaving a particular interface in the high priority queue From the BayRS Site Manager e Usea criteria path of Criteria Add IP IP UDP Destination Port e The range is 28000 to 28255 e The action path is Action IP Add High Queue Note This example shows how to give H 323 traffic priority over other protocols on the interface P0609327 02 Interoperability 181 Using VLAN on the network A virtual LAN VLAN is a logical grouping of ports controlled by a switch and end stations such as IP telephones configured so that all ports and end stations in the VLAN appear to be on the same physic
170. h call center on page 151 This system is less expensive and on a smaller scale However it does not offer PSTN fallback Figure 54 Connecting to IP telephones System telephone Central Office i2050 Software Phone Intranet VoIP trunk i2004 Home based users telephone or Call Center agents To set up this system 1 Ensure that each remote user has a network connection capable of supporting VoIP traffic such as DSL or cable On the Business Communications Manager set up the system to support IP telephones At the remote location install and configure an IP telephone Register each telephone and provide it with a DN a Ff o N Set up the DN record with the required lines and services P0609327 02 153 Appendix A Efficient Networking This section provides information about making your network run more efficiently Determining the bandwidth requirements on page 153 Network engineering on page 154 Additional feature configuration on page 159 Further network analysis on page 162 Post installation network measurements on page 165 Determining the bandwidth requirements The IP network 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 WAN requirements WAN must have enough capacity f
171. h range must end with an odd number You cannot have a total of more than 256 ports reserved Using a gatekeeper This section describes the use of a gatekeeper for your H 323 VoIP trunks For detailed information about the gatekeepers that the Business Communications Manager supports refer to the information under the headings below e Using Radvision ECS 3 2 GK as the gatekeeper on page 134 e Using CSE 1000 as a gatekeeper on page 136 e Gatekeeper call scenarios on page 141 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 address translation call control e admission control e bandwidth control zone management e IP registration A single Gatekeeper manages a set of H 323 endpoints This unit is called a Gatekeeper Zone A zone is a logical relation that can unite components from different networks LANS These Gateway zones such as the Business Communications Manager are configured with one or more alias names that are registered with the gatekeeper The gatekeeper stores the alias IP mapping internally and uses them to provide aliases to IP address translation services Later if an endpoint IP address changes that endpoint must re register with the gatekeeper The endpoint must also re register with the gatekeeper during the time to live TTL period if
172. he 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 Figure 63 Two calls on a full duplex link with silence compression BASE i O p Addu ON ete channeyink max ime Y Channel Link max DOMI ir Peak channel bandwidth is n peak Independent Tx and Rx bandwidth not bandwidth per call shared by half duplex calls Tx channel Bandwidth Comfort noise To provide a more natural sound during periods of silence comfort noise is added at the destination gateway when silence compression is active The source gateway sends information packets to the destination gateway informing it that silence compression is active and describing what background comfort noise to insert The source gateway only sends the information packets when it detects a significant change in background noise P0609327 02 173 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
173. his service is disabled Version digits Current version server software read only Description H 323 Voice Over The type of gateway available to H 323 terminals read only IP Gateway IP Telephony Configuration Guide 78 Chapter4 Installing NetVision telephones Assigning H 323 Terminals records The H 323 Terminals record Services IP Telephony IP Terminals identifies the NetVision handsets within the Business Communications Manager The Business Communications Manager uses the information from this file to determine if the handset will be allowed to connect to the system Pre configuration notes for NetVision handsets The following are some notes about the process of configuring handsets to the Business Communications Manager You must have an H 323 record configured before you configure the handsets with the Nortel NVPA Each telephone that you configure will use one IP client assignment so ensure that you added enough keycodes to accommodate both your IP telephones and your NetVision telephones The Name you specify in the H 323 record must match the User Name you specify in the Nortel NVPA tool otherwise the handset will not be allowed to connect to the Business Communications Manager If you do not specify a DN in the H 323 record one will automatically be assigned to the handset If you specified a DN record it will appear under the Active DNs heading once the handset connects to the system
174. hone prerequisites 40 IP telephones 49 network prerequisites 32 static IP list for NetVision 75 VLAN on IP telephones 50 181 VLAN site specific options 182 dialed digits VoIP trunk routing 114 dialing plan CDP 124 148 destination code and destination digits 117 118 destination digits 105 107 MI IPT prerequisite 148 outgoing calls 102 107 PSTN fallback 113 system prerequisites 34 UDP 148 Differentiated Service see DiffServ 199 DiffServ 199 DISA VoIP trunks 85 display keys configuration 48 Distributed Host Control Protocol see DHCP 53 DNs adding VoIP line pools 109 auto assign 34 auto assign IP telephones 51 before you start 75 changing handset name 82 H 323 terminals list 81 Hunt group target lines 98 NetVision 82 NetVision model 80 NetVision records 78 node range 122 records prerequisites 34 setting up target lines 98 documentation supporting 74 download firmware 65 staggered 66 DS30 split assessment 33 E E 164 94 echo cancellation 176 P0609327 02 Index 213 echo reply 173 efficient networking 153 Enable TTL 134 end to end delay 163 193 end to end DTMF signaling 176 Endpoint Type Radvision 135 end to end packet loss measuring 194 errors gathering statistics 196 network analysis 162 ethernet B W 154 155 156 ethernet connection IP telephones 47 external 117 118 F fallback activating VoIP schedule 119 configuring for PSTN 112 destination
175. hone to retry the connection to the Business Communications Manager VLAN 0 No VLAN 1 Manual VLAN 2 Automatically discover VLAN using DHCP Choose 0 NO VLAN if there is no VLAN on the network If you do not have DHCP on the network or if DHCP is supplied by a remote server select number 1 and enter the VLAN ID If you have the Business Communications Manager DHCP active on your system select number 2 if you want DHCP to automatically find the VLAN assignment Refer to Configuring DHCP on page 53 VLAN is a network routing feature provided by specific types of switches To find out if VLAN has been deployed on your System check with your network administrator If VLAN is deployed the system administrator responsible for the switch can provide the VLAN ID s for your system Refer to the Programming Operations Guide for information about VLAN configuration and DHCP Also refer to Using VLAN on the network on page 181 Cfg XAS 0 No default 1 Yes If you want the telephone to be able to connect to a Net6 service provider server choose 1 You will then be prompted for an IP address for the server Firewall note Ensure that the firewall filters are set up to allow IP traffic into and out of the system In the Programming Operations Guide refer to Configuring IP Firewall Filters for details P0609327 02 Chapter 3 Installing IP telephones 51 After you have entered all the
176. hony Set Sil C O IP Trunks H 323 Trunks seen ES SOMPIESSION i Media Parameters tab Set Jitter Buffer Medium Put 8 VoIP lines into the Pool O Services IP Telephony IP Trunks H 323 Trunks Local Gateway Interface IP Telephony Configuration Guide 124 Chapter6 Setting up VoIP trunks for outgoing calls Table 27 Fallback configuration for to create fallback between two systems Continued Task Settings for Santa Clara Settings for Ottawa Location in Unified Manager Give all system telephones access to the VoIP line pool Pool O Services Telephony Services System DNs Active set DNs Active Companion DNs and or All ISDN DECT DNs Line access Line pool access Confirm or assign target lines to all DNs or Hunt Groups that are assigned with the VoIP line pool lt targetline gt Services Telephony Services System DNs Active set DNs Active Companion DNs and or All ISDN DECT DNs Line access Line assignment Configure the target lines that you assigned Control set 3321 Control set 2221 Trunk Line data Line Type Private If busy To prime Services Telephony Services Lines Target lines Line XXX Prime set DN Prime set DN 3321 2221 Received number Received number 3322 2244 Create remote gateway Destination IP Destination IP Services IP Telephony record for remote 47 62 54 1 47 62 84 1 IP Trunks H 323 Trunks Communi
177. hooting Problem Suggested solution or cause Telephone does not connect to f an IP telephone does not display the text Connecting to server system within two minutes after power up the telephone was unable to establish communications with the Business Communications Manager Double check the IP configuration of the telephone and the IP connectivity to the Business Communications Manager cables hubs etc Slow connection between the If the connection between the IP client and the Business handset and the Business Communications Manager is slow ISDN dialup modem change the Communications Manager preferred CODEC for the telephone from G 711 to G 729 See IP telephone server configurations on page 49 One way or no speech paths Signaling between the IP telephones and the Business Communications Manager uses Business Communications Manager port 7000 However voice packets are exchanged using the default RTP ports 28000 through 28255 at the Business Communications Manager and ports 51000 through 51200 at the IP telephones If these ports are blocked by the firewall or NAT you will experience one way or no way speech paths Firewall note If you have the firewall filter set to Pass Outgoing and Block Incoming Except IP Phones this only allows IP telephony registration traffic through but blocks all other traffic including H 323 calls on this interface You must still specify an H 323 rule to allow IP call voice traffic
178. ic connectivity and basic telephony operations This guide assumes e You have planned the telephony and data requirements for your Business Communications Manager system e Operators have a working knowledge of the Windows operating system and of graphical user interfaces e Operators who manage the data portion of the system are familiar with network management and applications e The Business Communications Manager hardware is installed and initialized and the hardware is working External lines and internal telephones and telephony equipment are connected to the appropriate media bay modules on the Business Communications Manager e Keycodes for the required number of VoIP trunks IP telephones have been installed e If you are using VoIP trunks the keycode for MCDN has been installed e Configuration of lines is complete Refer to Chapter 2 Prerequisites checklist on page 35 for more information Symbols used in this guide This guide uses these symbols to draw your attention to important information Caution Caution Symbol Alerts you to conditions where you can damage the equipment Danger Electrical Shock Hazard Symbol AN Alerts you to conditions where you can get an electrical shock IP Telephony Configuration Guide 16 Preface gt Warning Warning Symbol Alerts you to conditions where you can cause the system to fail or work improperly 4 Note Note Tip symbol Al
179. ic in kbit s that IP telephony will put on the WAN From the table under Bandwidth requirements on full duplex links on page 156 the following figure shows the peak transmission rate for G 729 is 24 8 kbit s per call or 99 2 kbit s in each direction for all four calls In other words in order to support four G 729 calls the WAN link must have at least 99 2 kbit s of usable bandwidth in each direction The average bandwidth for each call is 12 4 kbit sec per channel or 49 4 kbit s for all four calls for each channel Low priority data applications can make use of bandwidth made available by silence suppression Figure 55 Peak traffic WAN link PPP B W Silence No SP Suppression peak peak Avg kbit s kbit s kbit s G 729 30 24 8 12 4 8 kb s P0609327 02 Efficient Networking 159 QoS Monitoring Bandwidth Requirement The VoIP Quality of Service QoS Monitor periodically monitors the delay and packet loss of IP networks between two peer gateways e g Business Communications Manager to Business Communications Manager by using a proprietary protocol The main objective of the QoS Monitor is to allow new VOIP calls to fall back to the PSTN if the IP network is detected as bad in terms of delay and packet loss For more details about configuring QoS Monitoring refer to the Business Communications Manager Programming Operations Guide The monitoring packets are delivered at
180. ide 216 Index VoIP trunk routing 114 link capacity insufficient 162 capacity system engineering 161 delay 163 full duplex bandwidth requirements 156 half duplex bandwidth requirements 155 local gateway Alias Names 94 Call Signaling 93 Fallback to Circuit Switched 93 Gatekeeper IP 94 Gateway Protocol 95 Registration TTL 94 Locating Server 51 MI IPT defined 25 gateway type 147 Interoperability 175 payload size 177 profile agreement 177 making calls VoIP trunks 125 Maximum cell rate MCR 154 MCDN gateway type 147 MI PT 25 MI IPT requirements 148 over VoIP 105 147 PRI fallback 148 remote gateway 147 Zone ID 144 measurements post installation 165 Measuring Intranet QoS 193 media channels asymmetrical negotiation 178 media parameters VoIP trunks 89 Media path redirection 175 media resources prerequisite 33 menu list feature 900 60 Meridian 1 MI PT 86 MCDN networking 147 profile 177 monitoring the network 165 MOS range 192 moving IP telephones 69 Keep DN alive 69 mu law 178 multi locations networking 151 MWI VoIP trunk interoperability 143 N name changing on handset 82 H 323 terminals list 81 H 323 Terminals record 80 NetVision 78 remote gateway 103 name display outgoing 144 NAT network prerequisites 32 Netmask IP telephones 49 network prerequisites 31 NetMeeting choosing media type 178 configuring clients 145 supports slowStart 178 NetVision before you
181. ifferent subnet you will need to make the appropriate changes to the telephone IP addressing However do not change the S1 IP address or the S2 IP address Note If your network is using partial DHCP reconfiguration is not required at this step Moving telephones and changing the DN To move a Nortel IP telephone and change the DN 1 Deregister the DN using the instructions in Deregistering DNs for IP telephones on page 66 Disconnect the network connection and the power connection from the telephone Reinstall the phone at the new location and reconfigure the telephone For information about this see Connecting the i200X telephones on page 47 IP Telephony Configuration Guide 70 Chapter 3 Installing IP telephones Configuring a new time zone on a remote telephone If the IP telephone connects to the system from a different time zone than the Business Communications Manager you can reset the telephone so that it displays the correct local time At the telephone enter FEATURE 510 Press CHANGE Press to toggle between and minus depending on which side of the time zone the telephone is located As a rule of thumb west is minus east is plus 4 Enter the number of hours difference 5 Press OK Offset time zones For areas such as Newfoundland Canada where the time zone is offset from a full hour press the key to add 5 to the number of hours then press OK Note The telephone is still
182. ign The purpose of planning is to design and allocate enough resources in the network to meet user needs QoS metrics or parameters help in meeting the needs required by the user of the service This section provides information about e Setting QoS on page 191 e Measuring Intranet QoS on page 193 e Implementing QoS in IP networks on page 198 e Network Quality of Service on page 200 Setting QoS There are two interfaces that must be considered when you set up QoS on the network as shown in the figure below e P telephony interfaces with the end users voice services made available need to meet user QoS objectives e The gateways interface with the intranet the service provided by the intranet is best effort delivery of IP packets not guaranteed QoS for real time voice transport IP telephony translates the QoS objectives set by the end users into IP adjusted QoS objectives The guidelines call these objectives the intranet QoS objectives IP Telephony Configuration Guide 192 Quality of Service Figure 64 Relationship between users and services Delay variation Business Communications Manager IP telephony parameters Fallback threshold Silence compression Codec Echo cancellation Non linear programming Business Communications Manager VoIP Corporate intranet Deliver voice fax service User oriented QoS Network QoS metrics Roundtrip conversation delay One way delay Clipp
183. iguring remote endpoints SIP trunks 200 e eee eee 106 Setting up the SIP address book i aac kae acicbackiecica daa debs 106 Outgoing call Gonna aoiceuebqa cd o Se Ede YOR e qd Pe ERE EI TEE FPE 107 IP Telephony Configuration Guide 6 Contents Managing H 923 and SIP TONKS ades inet nracevaemsdigaaeear er YET TR 107 Configuring lines and creating line pools llle 108 Configuring telephones to access the VoIP lines nunua ananunua 109 PSTN call to remote node 2 ais cent e dws exu ax edax mex x ud X EXE Ra eed 109 CB BIOCGDES 665446 d cce TR cC ora Waco n d o gU d ARN es 110 setting up VolP trunks for fallback isse suus Rer hk RR Rr xem e 112 Describing a fallback riGhWOIK 5222s m mmy m 113 Coniiguring routes for fallback ais aaepe Eoo b qox debi eed ub ice ius ews 114 P re contguradon requirements uascacssesssesc wer Y RE RR RR E RA 114 Adding routes Tor fallback 26x ace ddr EE EX QURERE CRX q E Xx iii 114 Assigning the line pools to routes lisi cee ceri nnn 115 Adding the destination code for the fallback route 5 116 Configuring the schedules for the destination codes 117 Setting up the VoIP schedule to overflow 00 0 eee eee 118 Activating the VoIP schedule for fallback 0 c cee eae eee 119 Deactivating the VoIP schedule 2 000 eee eee 119 How fallback routing WOIKS uua ss 564 s0h oe EREIGAGAQGGGERERXS NE ARR ORES 120 Exa
184. ing and dropout Packet loss echo Deliver IP service The IP gateway can monitor the QoS of the Intranet In this mode two parameters the receive fallback threshold and the transmit fallback threshold control the minimum QoS level of the intranet Fallback thresholds are set on pair per site basis The QoS level is aligned for user QoS metrics to provide an acceptable Mean Opinion Score MOS level The administrator can adjust the fallback thresholds to provide acceptable service to the users The settings in the following table indicate the quality of voice service IP telephony periodically calculates the prevailing QoS level per site pair based on the measurement of the following one way delay e packet loss e codec Table 50 Quality of voice service MOS Range Qualitative Scale MOS Range Qualitative Scale 4 86 to 5 00 Excellent 2 00 to 2 99 Fair 3 00 to 4 85 Good 1 00 to 1 99 Poor P0609327 02 Quality of Service 193 When the QoS level of any remote gateway is below the fallback threshold all new calls are routed over the standard circuit switched network if fallback is enabled The computation is taken from the ITU T G 107 Transmission Rating Model Measuring Intranet QoS Measure the end to end delay and error characteristics of the current state of the intranet These measurements help to set accurate QoS needs when using the corporate intranet to carry voice services The information u
185. ing screen appears On the Tools menu click Display Log The Mean Opinion Score Log File screen appears Close the browser window when you are finished viewing the log file P0609327 02 129 Chapter 7 Optional VoIP trunk configurations This section contains the procedures for configuring applications and features are not required on all networks or which are not Business Communications Manager products For details about setting up basic VoIP trunking refer to Chapter 5 Configuring local VoIP trunks on page 85 and Chapter 6 Setting up VoIP trunks for outgoing calls on page 101 This chapter contains information about e Port settings firewall on page 129 e Using a gatekeeper on page 133 e Faxing over VoIP lines on page 142 TP trunking interoperability settings on page 143 e Configuring NetMeeting clients on page 145 Port settings firewall In some installations you may need to adjust the port settings before the Business Communications Manager can work with other devices For details about setting port ranges refer to the information under the following headings e Using firewalls adding PortRanges on page 129 e Modifying PortRanges on page 131 e Port settings for legacy networks on page 132 Using firewalls adding PortRanges Firewalls can interfere with communications between the Business Communications Manager and another device The port sett
186. ing table describes the values for each display parameter Table 12 IP telephone server configurations Field Value Description DHCP 0 or 1 Enter 0 if your network is not using a DHCP server to dispense IP addresses Partial DHCP Enter 1 if your network does use a DHCP server If you choose to use a DHCP server rather than allocating static IP addresses for the IP telephones skip the remainder of this section For information about setting up DHCP server information for the IP telephones see Configuring DHCP on page 53 If DHCP 0 SET IP ip address The set IP must be a valid and unused IP address on the network that the telephone is connected to NETMASK subnet mask This is the subnet mask This setting is critical for locating the address system you want to connect to DEF GW ip address Default Gateway on the network i e the nearest router to the telephone The router for IP address W X Y Z is usually at W X Y 1 If there are no routers between the telephone and the Business Communications Manager network adaptor to which it is connected for example a direct HUB connection then enter the Published IP address of the Business Communications Manager as the DEF GW If the IP telephone is not connected directly to the Published IP address network adaptor set the DEF GW to the IP address of the network adaptor the telephone is connected to For information on setting
187. ings 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 A Nortel Networks IP telephone uses ports between 51000 and 51200 to communicate with the Business Communications Manager The Business Communications Manager by default uses ports 28000 to 28255 to transmit VoIP packets IP Telephony Configuration Guide 130 Chapter 7 Optional VoIP trunk configurations Follow these steps to add a port range 1 In Unified Manager open Services IP Telephony Port Ranges 9 IP Telephony System Configuration IP Terminals IP Trunks a 2 From the top menu click Configuration and then select Add PortRanges Modify PortRanges Add PortRanges Delete PortRanges The PortRanges dialog box appears Refer to Figure 42 Figure 42 Port ranges dialog box PortRanges PortRange RH Format Rit Beain n End n 3 Enter the port settings Table 29 Media parameters record Field Value Description PortRange R read only This field indicates the range of ports that are available for this application Begin range This indicates the first port setting in the range 1024 65534 gt End lt range This indicates the last port setting in the range 1025 65535 gt P0609327 02 Chapter 7 Optional VoIP tr
188. inued Prerequisites Yes 1 e Answer this only if your system will use a gatekeeper otherwise leave it blank Does the network diagram contain the IP address for any Gatekeeper that may be used Note If the network has a Meridian 1 running IPT software you cannot use a RadVision gatekeeper Network devices The following table contains questions about devices on the network such as firewalls NAT devices and DHCP servers e Ifthe network uses public IP addresses complete 2 d e If the network uses private IP addresses complete 2 e to 2 f Table 2 Network device checklist Prerequisites Yes No 2 a Is the network using DHCP 2 b If so are you using the DHCP server on the Business Communications Manager 2 c Is the network using private IP addresses 2 d Are there enough public IP addresses to accommodate all IP telephones and the Business Communications Manager 2 e Does the system have a firewall NAT device or will the Business Communications Manager be used as a firewall NAT device NOTE NetVision handsets do not work on a network that has NAT between the handset and the system 2 f If the Business Communications Manager is to be used as a firewall NAT device do the firewall rules fit within the 32 input rules and 32 output rules that the Business Communications Manager provides 2 g A hub based core will not have suitable performance for IP Telephony Does the net
189. ion and NetVision Data telephones use H 323 Refer to Chapter 4 Installing NetVision telephones on page 73 The applications that control these protocols on the Business Communications Manager provide an invisible interface between the IP telephones and the digital voice processing controls on the Business Communications Manager The i200X IP telephones are hardwired to an internet connection They can be installed on any internet connection that has access to the network connected to the LAN or WAN of the Business Communications Manager The Nortel Networks 12050 Software Phone runs on any computer running Windows 98 Windows 2000 or Windows XP The computer must be connected to the LAN or WAN to which the Business Communications Manager is connected Configuring Nortel Networks i series telephones The configuration menus for the Nortel Networks IP telephones are located under Services IP Telephony Nortel IP Terminals and Services Telephony Services System DNs Inactive DNs Set DNs DN records move to Active set DNs after the telephone connects registers to the system Once a DN record is assigned and the telephone registers to the system the record also shows up under DN Registration in one of the following folders e Inactive DNs reg d if the IP telephone has been assigned a DN and is registered to the system but currently is not active e IP set DNs reg d if the IP telephone is active e IP wireless DNs reg d if the NetVis
190. ion handset is registered and is active The information under the following headings provides details about configuring and troubleshooting IP telephones e Preparing your system for IP telephone registration on page 43 e Installing i series telephones on page 46 e Configuring the i20XX telephones to the system on page 47 e Troubleshooting IP telephones on page 51 P0609327 02 Chapter 3 Installing IP telephones 43 Preparing your system for IP telephone registration When you install an IP telephone on a Business Communications Manager you must activate terminal registration on the Business Communications Manager If this is your first installation you need to set the general parameters for IP registration For the simplest installation possible set telephone Registration and Auto Assign DNs to ON and leave the Password field blank IP telephones installed on the system LAN will connect and boot up without manual registration Security Note Turn Registration and Auto Assign DNs off once the telephone s are e registered Nortel cautions that leaving your IP registration open and unprotected by a password may pose a security risk Caution DN auto assign Call Center DNs CDNs and IP telephones share a common DN database If you use auto assign to assign DNs to your IP telephones the system can inadvertently assign an IP telephone to a DN within the CDN range CDNs do not register to the syste
191. ion 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 The following figure shows the peak bandwidth requirements for one call on a full duplex link with silence compression enabled The spare bandwidth made available by silence compression is used for lower priority data applications that can tolerate increased delay and jitter Figure 62 One call on a full duplex link with silence compression Tx Hello Fred This is Susan Do you have a minute Fred here Hi Sure Pd Bandwidth used X Channel Link max Conversation me d Channel Link n sl Time Indep ndent Tx and Rx bandwidth not shared by half duplex calls Bandwidth available for data apps Rx channel Bandwidth Tx channel Bandwidth IP Telephony Configuration Guide 172 Silence compression When several calls are made over a full duplex link all calls share the same transmit path and they share the same receive path Since the calls are independent the peak bandwidth must account for the possibility that all speakers at one end of the link may talk at the same time Therefore the peak bandwidth for n calls is n the full transmission rate The following figure shows the peak bandwidth requirements for two calls on a full duplex link with silence compression Note that the peak bandwidth is twice the full transmission rate even though t
192. is on a call when the diversion command was issued e Once hot desking occurs between two IP telephones no activity is allowed on the originating telephone except to cancel hot desking The display on the originating telephone indicates where it has been diverted On the diverted telephone the key displays reflect the displays from the originating telephone e Call forwarding to voice mail continues as normal Voice mail can be accessed from the diverted IP telephone as if it were the originating telephone e When hot desking is cancelled which can be performed from either telephone the displays for each telephone return to normal Note You must wait 10 seconds after completing a call before you cancel hot desking e Using headsets with hot desking If you use the headset feature on your IP telephone and you want to hot desk your telephone to another IP telephone ensure that the target telephone also a headset installed before you enter the hot desking feature e Ifthe target telephone does not have a headset the headset feature from the first telephone does not transfer during hot desking Adding a headset to the target telephone after hot desking is enabled does not correct the situation To enable headset function in this situation you need to cancel hot desking plug a headset into the target telephone and then re establish hot desking P0609327 02 Chapter 3 Installing IP telephones 63 e Ifthe target telephone has a h
193. istering the telephone to the system IP Telephony Configuration Guide 48 Chapter3_ Installing IP telephones Registering the telephone to the system When you first connect the telephone to the IP connection you may receive one of the following e If the telephone is not yet registered and if a password was entered in the Terminal Registration screen the telephone prompts you for that password e If you set Auto Assign DN to OFF the telephone prompts you for a DN Refer to Setting IP terminal general settings on page 43 e If you are prompted for a password enter the password and press OK e If you are prompted for a DN enter the DN you want assigned to this telephone and press OK When the telephone registers it downloads the information from the Business Communications Manager IP Telephony record to the telephone configuration record This might include a new firmware download which occurs automatically If new firmware downloads the telephone display indicates the event Note If the telephone displays a prompt that indicates it cannot find the server follow the instructions in Configuring telephone settings to enter the specific network path Troubleshooting IP telephones on page 51 describes other possible prompt messages After registration is complete you do not need to go through the registration steps described above unless you deregister the terminal For information about setting the registration set
194. l H 323 Terminal List General Default Codec 8 723 v G 729 G 711 uLaw G 711 aLaw 4 Use the information in the table below to determine default codec settings Table 18 H 323 Terminal list Field Value Description Codec G 729 Specifying a non default CODEC for a telephone allows you G 711 uLaw to override the general setting You might for example want to specify a low bandwidth CODEC like G 729 for a telephone G 711 aLaw that connects to a remote or busy sub net P0609327 02 Chapter 4 Installing NetVision telephones 77 Monitoring H 323 service status The Summary screen under H 323 terminals tells you what connection status is available to H 323 terminals 1 Inthe Unified Manager click the keys beside Services IP Telephony and IP Terminals 2 Click H 323 Terminals The Summary screen is the visible tab Figure 18 Viewing the Summary tab for H 323 terminals Summary General H 323 Terminal List Summary Name VolP Gateway Status Up v Version 35 70 0 33 Description 1 323 Voice Over IP Gateway 3 The following table describes the fields on the screens Table 19 H 323 terminals Summary fields Field Value Description Name VoIP gateway This is the type of gateway that the H 323 handsets will be using read only Status Up UP H 323 handsets can be administered on this system Enabled Enabled This service is enabled Disabled Disabled T
195. lPilot before the fax machine s timer is started P0609327 02 Chapter 7 Optional VoIP trunk configurations 143 e The call duration can be increased by adding a timed pause to the end of dialing string for example 758 5428 This allows the call to ring at the destination before the fax machine call duration timer starts e Since the problem is related to the delay in initiating the fax session the number of rings for fax mailboxes Call Forward No Answer CFNA should be minimized IP trunking interoperability settings The IP trunking heading provides interoperability settings for VoIP trunks for CLID transmission MWI from a remote system and private networking identification numbers that are required for some types of system servers If the VoIP network has interoperability issues with the end point system these settings can be adjusted to accommodate the requirements of the other system To access these settings 1 Inthe Unified Manager press the keys beside Services and General Settings 2 Select IP trunking Figure 47 IP trunking interoperability fields General settings IP trunking Feature settings Nortel IP terminals Send Name Display Remote Capability MWwI Virtual Private Network Id Zone Id Y r r 3 The following table describes the field properties for each item Table 38 IP trunking interoperability fields Field Value Description support MWI
196. line For more information about MOS refer to Appendix E Quality of Service on page 191 and the QoS Monitor chapter of the Programming Operations Guide Gateway Type BCM3 6 Choose the type of system that is accessed through the BCM3 5 remote gateway BCM3 0 BCM3 6 Business Communications Managers running 3 6 BCM2 5 software BCM2 0 BCM3 5 Business Communications Managers running 3 5 software CSE 1000 BCM3 0 Business Communications Managers running 3 0 CS 2000 software IPT BCMA2 5 Business Communications Managers running 2 5 NetMeeting or 2 5 FP1 or FP1 Maintenance Release software Norstar IP BCM 2 0 Business Communications Managers running 2 0 Gateway software or Enterprise Edge systems running 2 0 x software Other IPT M1 Internet Telephony Network Gateway CSE 1000 CSE1000 switch CS3000 CS3000 switch NetMeeting Microsoft NetMeeting trunk protocol Norstar IP Gateway Norstar IP trunk protocol If your gateway is set to BCMX X and the other system is upgraded to 3 6 your system will automatically update this listing to BCM3 6 when the other system is contacted after the upgrade If this does not occur your original configuration may not be correct and you will have to set the change manually Gateway Protocol None Select the gateway protocol that the trunk expects to use SL 1 None No special features CSE SL 1 Use for BCM 2 5 systems only that require MCDN over VoIP trunks CSE MCDN protocol for gateways that provi
197. m so the system has no way of knowing that the DN is already assigned If this happens you can rename the IP DN to a DN outside the CDN range and reenter the CDN information Setting IP terminal general settings 1 In Unified Manager open Services IP Telephony IP Terminals and Nortel IP Terminals 2 Select the General tab The General screen appears Figure 5 Set registration properties 9 IP Telephony Summar General IP Terminal Status System Configuration T IP Terminals r General H 323 Terminals Nortel IP Terminals Fegitiation Password Auto Assign DNs Advertisement Logo Default Codec AUTO Default Jitter Buffer AUTO G 729 Payload Size ms 30 G 723 Payload Size ms 30 G 711 Payload Size ms 20 IP Telephony Configuration Guide 44 Chapter 3 Installing IP telephones 3 Use the information in the table below to set up your IP terminals general information Table 11 IP terminals general record fields Field Value Description Registration On Set this value to ON to allow new IP clients to register with Off the system WARNING Remember to set Registration to Off when you have finished registering the new telephones Password lt 10 alphanumeric gt This is the password the installer will enter on the IP Default bcmi telephone to connect to the Business Communications Manager If this field is le
198. m the deletion Select an Option x Are you sure you want to delete this row Yes No Cancel Waring Applet Window Click the Yes button to delete the record P0609327 02 Chapter 4 Installing NetVision telephones 83 Under the Systems DNs heading the DN record returns to the Inactive DNs list and disappears from the DN Registration lists Deregistering a telephone If you want to deregister a Net Vision handset you use the DN registration record 1 In the Unified Manager click the keys beside Services System DNs DN registration IP wireless DNs reg d Click the key beside one of the following Active if you are deregistering an active NetVision handset Inactive if you are deregistering an inactive NetVision handset Select the DN for the NetVision handset you want to deregister Click on the Configuration menu then select Deregister Refer to the figure below If you run Deregister on an active device you will be prompted to confirm that you understand that the device will be terminated If you click OK the device is deregistered immediately If you run Deregister on an inactive device there will be no prompts and the action will occur immediately Figure 21 Deregister DN from Configuration menu 9 IP wireless DNs red d Ad Registration type iP wireless IP Telephony Configuration Guide 84 Chapter4 Installing NetVision telephones P0609327 02 85
199. machine Here are some suggestions to minimize the possibility of your VoIP calls being dropped because of fax tone interference Caution Operations note Fax tones that broadcast through a telephone speaker will e Locate fax machine away from other telephones e Turn the speaker volume on the fax machine to the lowest level or off Fax tones recorded in a voice mailbox In the rare event that fax tones are captured in a voice mail message opening that message from an telephone using a VoIP trunk will cause the connection to fail For a list of limitations and requirements for using T 38 fax refer to T 38 fax restrictions and requirements on page 189 Operational notes and restrictions Some fax machines will be unable to successfully send faxes over VoIP T 38 trunks to the following destinations CallPilot mailboxes e CallPilot mailboxes accessed through auto attendant e Fax Transfer calls transferred to a system fax device through the auto attendant Use the following tips to avoid this problem e Avoid the use of manual dial on the originating fax machine In some fax machines manually dialing introduces a much shorter call time out e If manual dial must be used then the user should wait until the call is answered before starting the fax session e If manual dial must be used then the user should enter the digit 8 before initiating the fax session This ensures that the fax session is initiated by Cal
200. minals record for each handset you install 1 Inthe Unified Manager click the keys beside Services IP Telephony and IP Terminals 2 Click H 323 Terminals 9 Q IP Telephony System Configuration 9 IP Terminals P Nortel IP Terminals 3 Onthe top menu click Configuration and choose Add Entry Add Entry Update Entry Delete Entry The H 323 Terminal List dialog box appears Figure 19 H 323 Terminal list dialog box H 323 Terminal List Name Format Unique across first 7 chars DN pg Password 1534 IP Address Status Codec Default v 4 Use the information in the table below to set up your NetVision handset IP system record IP Telephony Configuration Guide 80 Chapter 4 Installing NetVision telephones Table 20 H 323 Terminal list Field Value Description Name lt alphanumeric gt This is the name for the handset This name must have unique characters for at least the first seven digits Note This is the same name that you will enter in the Nortel NVPA configuration record for the User Name of the handset This name must be unique within the first seven characters for each handset and can be a maximum of 10 characters DN DN number or This is the assigned DN for this handset If you want the 0 system to dynamically define a DN enter O zero Note This field cannot be left blank Password numeric Enter a unique password This is what the
201. mple A private network configured for fallback llus 122 System programming for networking and fallback routes 123 Making calls through a private VoIP network gateway 125 Connecting an i200X telephone anaana naaa 125 PSTN falak Wed sous 6658 owen caegesreees EAT teens nates SG Qa Rid 126 Fesetting ilie NOG uus cuu iR ORI xh S e e EY aoe 127 Civility of Service RIDE 1iuiuc aed uomo Ide e det ok eae ad cb ee d ec UA Uc Aa 127 Quality Of Semice SAUE s ons a ERE RERESSRERENSRERaeErRPREEdOSGRd p REA 127 Updating the QoS monitor data ssascssss edt e Y RE REREY RR ERA 128 Viewing QoS monitoring logging s xsssesosua esu i RR Y ETELRSER 3G 128 Chapter 7 Optional VoIP trunk configurations 0000 e eee eee eee 129 Por setings CINCY ac aoousestca ask toned dad eda aU aa oe spied anion oe Rok 129 Using firewalls adding PortRanges cece eee eee ee 129 Moding PortRangaS MR TER PE EE 131 Port settings for legacy networks lees 132 BLU DENecUA c P P MVUEET 133 Using Radvision ECS 3 2 GK as the gatekeeper 0 0 2 c eee eee eee 134 Configuring Radvision for Business Communications Manager 134 Using CSE 1000 as a gatekeeper ccisseiisvase har RR EEG A Rx RA ves 136 Business Communications Manager requirements 20 0 005 136 CSE 1000 configuration adding an H 323 endpoint 0 137 Setting the
202. n the DiffServ field which was known as the ToS field in older versions The Business Communications Manager assigns Expedited Forwarding EF PHB for voice media packets and the Class Selector 5 CS5 PHB for voice signaling control packets On the Business Communications Manager these assignments cannot be adjusted The Business Communications Manager system performs QOS routing but if one or more routers along the network route do not support QOS routing this can impact voice quality Business Communications Manager 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 30 Chapter1 Introduction P0609327 02 31 Chapter 2 Prerequisites checklist Before you set up voice over IP VoIP trunks or IP telephones on a Business Communications Manager complete the following checklists to ensure that the system is correctly set up for IP telephony Some questions do not apply to all installations This guide contains a number of appendices that explain various aspects of IP networking directly related to IP telephony functions Refer to the Programming Operations Guide for specific information about configuring the data portion of the Business Communications Manager This section includes the following checklists e Network diagram on page 31 e Network devices on page 32 e Network assessment on page 3
203. nction exchanges UDP probe packets between all monitored gateways to collect the network statistics for each remote location All the packets make a round trip from the Sender to Receiver and back to the Sender From this information you can calculate the latency and loss in the network for a distinct location Note 1 Quality of Service monitoring is supported only on Business Communications Manager MI with IPT 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 P0609327 02 Quality of Service 201 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 e 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
204. nd of structure iii jjj kkk 111 ppppp identifies IP port for server ASCII encoded decimal P0609327 02 Chapter 3 Installing IP telephones 55 aaa identifies Action for server ASCII encoded decimal range 0 255 rrr identifies retry count for Business Communications manager ASCII encoded decimal range 0 255 This string may be NULL terminated although the NULL is not required for parsing Notes aaa and rrr are ASCII encoded decimal numbers with a range of 0 255 They identify the Action Code and Retry Count respectively for the associated Business Communications Manager Internal to the IP telephones they will be stored as 1 octet 0x00 0xFF Note that these fields must be no more than three digits long the Business Communications Manager is always considered the Primary server the second server always considered Secondary if only one Business Communications Manager is required terminate primary TPS sequence immediately with instead of e g Nortel i2004 A iii jjj kkk lll ppppp aaa rrr valid options are one Business Communications Manager or two Business Communications Managers 0 3 not allowed Action code values 0 reserved UNIStim Hello currently only this type is a valid choice 2 254 reserved 255 reserved iii jjj kkk 111 are ASCII encoded decimal numbers representing the IP address of the Business Communications Manager They do not need to be three digits long as the and
205. nd 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 IP Telephony Configuration Guide 166 Efficient Networking P0609327 02 167 Appendix B Silence compression This section describes using silence compression on half duplex and full duplex links e Silence compression on half duplex links on page 168 e Silence compression on full duplex links on page 170 e Comfort noise on page 172 Silence compression reduces bandwidth requirements by as much as 50 per cent This section explains how silence compression functions on a Business Communications Manager network For information about enabling silence compression in VoIP gateways refer to Configuring media parameters on page 89 G 723 1 and G 729 Annex B support Silence compression A key to VoIP Gateways in business applications is reducing WAN bandwidth use Beyond speech compression the best bandwidth reducing technology is silence compression also known as Voice Activity Detection VAD Silence compression technology identifies the periods of silence in a conversation and stops sending IP speech packets during those periods Telco studies show that in a typical telephone conversation only about 36 to 40 of a full duplex conversation is active When one person talks the other listens This is half duplex There are important p
206. nder the following headings provides more details about measuring QoS e Measuring end to end network delay on page 193 e Measuring end to end packet loss on page 194 e Recording routes on page 194 e Adjusting Ping measurements on page 195 e Measurement procedure on page 196 Measuring end to end network delay The basic tool used in IP networks to get delay measurements is the Ping program Ping takes a delay sample by sending a series of packets to a specified IP address and then returning to the originating IP address Ping then displays statistics for the packets High packet times can 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 The file can be analyzed
207. ndpoint choose name from list This is the name of another H 323 endpoint Picking a name in this field provides a tandem endpoint 5 Click Create H323 Setting the H 323 Endpoint Dialing Plan AII dialing plan information must be consistent on all H 323 endpoints using the gatekeeper Follow these steps to set the dialing plan into the Gatekeeper Admin tool 1 Select GK Standby DB Admin 2 Select NumberPlanEntries 3 Select Create 4 Ensure that the Endpoint you select is the one for which you want to create a numbering plan entry 5 Click Select IP Telephony Configuration Guide 138 Chapter 7 Optional VoIP trunk configurations 6 Ensure that the following fields are set Table 35 CSE 1000 H 323 dialing plans Field Value Description Number digits This is the unique number that identifies the Business Communications Manager Type choose from list This is the TON Type of Number or NPI Numbering Plan Identifier for the endpoint EntryCost digits 1 255 gt This value determines which destination the gatekeeper will deliver to if the leading digits are the same for more than one endpoint The gatekeeper will select the endpoint with the lowest EntryCost value 7 Click Create Committing Gatekeeper Configuration Changes Gatekeeper changes occur in the standby database For these settings to be used by the active gatekeeper you must
208. ne Administrator Pulse Code Modulation Packet InterNet Groper Power inline patch panel Point to Point Protocol Primary Rate Interface Public Switched Telephone Network Quality of Service Registration Admissions and Status Real time Transfer Protocol Session Initiation Protocol Simple Network Management Protocol Transmission Control Protocol User Datagram Protocol or Universal Dialing Plan UNISTIM Terminal Proxy Server Voice over Internet Protocol Voice Activity Detection Virtual LAN Wide Area Network Related publications Documents referenced in the JP Telephony Configuration Guide include Installation and Maintenance Guide Software Keycode Installation Guide e Programming Operations Guide Telephony Feature Handbook e 1200X and 12050 Software Phone user cards P0609327 02 Preface 19 How to get help If you do not see an appropriate number in this list go to www Nortelnetworks com support USA and Canada Authorized Distributors ITAS Technical Support Telephone 1 800 4NORTEL 1 800 466 7835 If you already have a PIN Code you can enter Express Routing Code ERC 196 If you do not yet have a PIN Code or for general questions and first line support you can enter ERC 338 Website http www nortelnetworks com support Presales Support CSAN Telephone 1 800 4NORTEL 1 800 466 7835 Use Express Routing Code ERC 1063 EMEA Europe Middle East Africa Technical Support CTA
209. ng a half duplex conversation The following figure shows the peak bandwidth requirements for one call on a half duplex link with silence compression enabled Because the sender 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 59 One call on a half duplex link with silence compression Tx Hello Fred This is Susan Do you have a minute Fred here Hi Sure Rx Conversation P d Bandwidth used Channel Link max Rx Chan vcn Time Bandwidth Tx Half duplex call alternates use of half duplex bandwidth P0609327 02 Silence compression 169 Figure The effect 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 be aggregated for a number of calls The following figure shows the peak bandwidth requirements for two calls on a half duplex link with silence compression enabled The peak bandwidth for all calls is equal to the sum of the peak bandwidth for each individual call In this case that is twice the full transmission rate for the two calls 60 Two calls on a half duplex link with silence compression Conversation Tx Buenos noches Juan Mu
210. ng Bandwidth Requirement on page 159 Engineer the network for worst case numbers to indicate the spare bandwidth a LAN must have to handle peak traffic It is important to plan so that the LAN WAN can handle the IP telephony traffic using the defined codec without delay or packet loss The installer or administrator must select one configuration and then set up the LAN WAN so there is more bandwidth than the IP telephony output The following table provides bandwidth characteristics for the transmission of voice over IP for various link types given codec type and payload sizes The bandwidths provided in this table explain the continuous transmission of a unidirectional media stream Table 38 VoIP Transmission Characteristics for unidirectional continuous media stream Codec Type Payload Size IP Packet Ethernet B W PPP B W FR B W ms Bytes Bytes kbit s kbit s kbit s G 711 64 kb s 30 240 280 81 6 75 2 77 1 G 729 8 kb s 30 30 70 25 6 19 2 21 1 G 723 1 6 3 kb s 30 24 64 24 0 17 6 19 5 G 723 1 5 3 kb s 30 20 60 22 9 16 5 18 4 Notes 1 indicates payload sizes used by Business Communications Manager 3 5 for transmission Other values listed indicate payload sizes that the Business Communications Manager 3 5 can receive 2 Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter frame gap P0609327 02 Efficient Networking 155 The peak ban
211. ng local VoIP trunks 99 Configuring target lines There are two places where target lines need to be configured The target line is assigned to a telephone or Hunt group by assigning a free target line 241 to 492 to the telephone DN record or Hunt group The incoming digits e g 3321 are assigned to the target line the same one you assigned to the telephone by setting the Received Number under that target line to the incoming digits If your system does not have target lines already assigned use this procedure to assign target lines to individual telephones Note You can also use the Add Users wizard if you need to create target lines for a range of telephones Refer to the Programming Operations Guide for detailed information about using the wizard 1 In Unified Manager open Services Telephony Services System DNs 2 Under the Active Set DNs Active Companion DNs or All ISDN DECT DNs or under the Inactive DNs if you are preconfiguring DN records choose the DN record of the telephone where you want the line to be directed 3 Choose Line Access Line assignment and click the Add button 4 Inthe Line field enter the number of an available target line 241 492 5 Click the Save button 6 Click the line number you just created and ensure that you have the line set to Ring Only if the telephone has no line buttons set for the line or Appearance and Ring if you are adding this to a DN that has line keys or which will be
212. nloads nvfirmware2 html The serial cable required to update the programming of the handset can be purchased from Purchased from Symbol at lt http symbol com gt part number 25 20528 01 packet Group of bits transmitted as a complete package on a packet switched network packet switched network PSTN A telecommunications network based on packet switching technology A link is busy for the duration of the packets Ping This utility is used to echo messages to a host over an IP network This allows you to find out if the other point is available Ping also can include statistics about how long it took from end to end which provides information about routing prioritization This refers to how a voice data packet is set up in the Business Communications Manager so that external routers recognize it as having a high priority thus shortening delay times and increasing the perception of voice quality over VoIP trunks published IP address The IP address that both the IP telephones and the Symbol NetVision telephones use to access the Business Communications Manager NetVision uses the H 323 RAS protocol QoS quality of service routing To minimize voice jitter over low bandwidth connections the Business Communications Manager programming assigns specific DiffServ Marking in the IPv4 header of the data packets sent from IP telephones During the packet journey through the network including any routers on that network the
213. nnection to a Meridian system For details about setting up MCDN networks refer to the Private Networking chapter in the Business Communications Manager Programming Operations Guide Note If you use MCDN over VoIP ensure that your PSTN fallback line is a PRI SL 1 line to maintain MCDN features on the network One application of this type of network might be for a company which has an M1 at Head Office who want to set up a warehouse in another region This would allow the warehouse to call Head Office across VoIP lines bypassing long distance tolls This type of network also provides the possibility of having common voicemail off the M1 Refer to the following figure for an example IP Telephony Configuration Guide 148 Chapter8 Typical network applications using MCDN Figure 50 M1 to Business Communications Manager network diagram Telephone i mea cpi PSTN Head Office Warehouse M1 IPT Business Communications Manager fallback route System telephone Intranet VoIP trunk Company server telephone To set up this system 1 Make sure the M1 IPT meets the following requirements IPT version 3 0 or newer 2 Ensure that the MI ESN programming CDP UDP is compatible For information about this refer to your M1 documentation 3 Onthe Business Communications Manager Unified Manager Set up outgoing call configuration for the VoIP gateway Set up a remote gatewa
214. nning BCM 3 5 or newer software If you need more information contact your technical support service Counting IP trunks After you enter the keycodes for your VoIP trunks you need to specify how many of the trunks will be used for H 323 trunking and how many for SIP trunking Once these parameters are set you can go to Line programming Services Telephony Services to determine the parameters for each line including assigning line pools for each type of trunk so that you can configure the routing You must also assign the H 323 or SIP line pools to each telephone that you want to be able to call out over the lines Since H 323 trunks and SIP trunks use the same pool of available lines 001 to 060 you can use the IP Trunks Settings screen to keep track of the total number of enabled trunks and how they are distributed between H 323 and SIP trunks Determining the IP trunk count 1 Enter the keycodes that you need to enable enough IP lines for your requirements 2 In Unified Manager click the keys beside Services IP Telephony 3 Click IP Trunks The IP Trunks Settings screen appears Figure 22 IP Trunks Settings screen IP Trunks Settings 9 IP Trunks 9 H 323 Trunks Remote Gateway Maximum Trunks jo Q SIP Trunks Address Book Total Trunk Credits p Number of H 323 Trunks 2 Number of SIP Trunks 2 The first two fields are read only and are determined by the number of IP trunk key
215. nsider 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 is enabled The Ethernet LAN is half duplex Ethernet LAN may also be full duplex Given the above what is the peak traffic in kbit s that IP telephony will put on the LAN From the table under Bandwidth requirements on half duplex links on page 155 the following figure shows the peak transmission bandwidth for G 729 with silence suppression enabled on a half duplex link is 34 4 kbit s per call or 137 6 kbit s for all four calls Figure 54 LAN engineering peak transmission Ethernet B W Silence NoSP Suppression peak peak Avg kbit s kbit s kbit s G 729 30 34 4 34 4 8 kb s IP Telephony Configuration Guide 158 Efficient Networking WAN engineering Wide Area Network WAN links are typically full duplex links both talk and listen traffic use separate channels For example a T1 link uses a number of 64 kbit s DSO duplex channels allowing 64 kbit s for transmit path and n 64 kbit s for the receive path WAN links may also be half duplex Example 1 WAN engineering voice calls Consider a site with four IP telephony ports and a full duplex WAN link using PPP The preferred codec is G 729 kbit s which uses a voice payload of 20 ms Silence compression is enabled Given the above what is the peak traff
216. nt for processing The Ping measurements are taken from Ping host to Ping host The Transmission Rating QoS metrics are from end user to end user and include components outside the intranet The Ping statistics for delay requires additional adjustments by adding 140 ms to explain the processing and jitter buffer delay of the gateways No adjustments are required for error rates If the intranet measurement barely meets the round trip QoS objectives the one way QoS is not met in one of the directions of flow This state can be true when the flow is on a symmetric route caused by the asymmetric behavior of the data processing services Late packets Packets that arrive 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 IP Telephony Configuration Guide 196 Quality of Service Measurement procedure The following procedure is an example of how to get delay and error statistics for a specific site pair during peak hours Program a script to run the Ping program during the intranet peak hours repeatedly sending a series of 50 Ping requests Each Ping request
217. ny including T 38 fax cannot operate on a system that has PEC Is installed on the MSC Your system must have PEC IIIs Table 4 Resource assessment Prerequisites Yes No 4 a Has a Business Communications Manager Resource Assessment been performed using the resource questionnaire in the Programming Operations Guide 4 6 Has an analysis been done to determine which DS30 split is appropriate for the system Has the DS30 split been changed to 3 5 if necessary 4 c Have all necessary media resources for IP trunks clients vmail IP music or WAN dialup been assigned or dedicated IP Telephony Configuration Guide 34 Chapter2 Prerequisites checklist Keycodes All elements of VoIP trunks and IP telephony are locked by the Business Communications Manager keycode system You can purchase keycodes for the amount of access you want for your system Additional keycodes can be added later providing there are adequate resources to handle them Table 5 Keycodes Prerequisites Yes No VoIP keycodes Both H 323 trunks and SIP trunks use VoIP keycodes 5 a Complete this question only if you are using VoIP trunks Do you have enough registered it occupies an IP client whether it is active or not 5 b Complete this question only if you are using IP telephones Do you have enough IP client keycodes Note IP clients and IP telephones are a 1 1 ratio Include any NetVision teleph
218. o Business Communications Manager systems if both systems are running BCM version 3 5 or newer software 3 0 1 and prior Dialing protocols MCDN networks Do not support the M1 requirement for specific tags for Local National and International calls tandemned over a Business Communications Manager network to the public network 3 0 1 and prior Does not support the T 38 fax protocol IP Telephony Configuration Guide 186 Interoperability Table 47 Software interoperability restrictions and limitations for IP trunking Continued Software release Description of restriction limitation 3 0 1 and prior The profile on the IPT must be set to the same first preferred codec as that of the Business Communication software IPT card must be version 3 0 or 3 1 In order for features such as Transfer and Conference to operate correctly between all Business Communications Managers and IPTs in a network these are the rules The First Preferred Codec for VoIP Trunks must be the same on all Business Communications Managers This is configured in Unified Manager under Services IP Telephony H 323 Trunks Media Parameters In addition if the first preferred codec is G 729 or G 723 the Silence Suppression option on that page must be the same on all Business Communications Managers in the network The Business Communications Manager supports only basic call to from NetMeeting S W version FP1 GA 3
219. o use the features list Using the Services button to access features The IP telephone has a limited number of memory buttons that can be configured with lines or features however a soft features menu also can be accessed by pressing the Services button Ce e Use the up and down directional buttons or the Page and Page display keys to move quickly through the list e Press the Select display key to activate the feature then use the feature as you normally would For example if you selected Call Forward enter the number you to which you want to forward the call Or if you select speed dial FEATURE 0 enter the speed dial code for the number you want the telephone to dial This feature allows you to assign your hardware feature keys to line and intercom applications and still access the Business Communications Manager call features without needing to remember a feature code Although the list is defaulted to the Services button you can assign the display list to one of the other hard feature keys The user can also assign it as a memory button using FEATURE 3 at a specific telephone Refer to the Programming Operations Guide for information about programming IP telephone memory buttons under User Preferences Note If you move the feature to another memory button the Services button no longer accesses the menu P0609327 02 Chapter 3 Installing IP telephones 61 Resetting the Hot Desking password You can tran
220. oIP calls The incremental peak bandwidth for VoIP traffic is therefore R1 R2 peak VoIP Load 4 28 kbit s 112kbit s R4 R5 peak VoIP Load 8 28kbit s 224kbit s With Business Communications Manager VoIP gateway bandwidth requirements and Traceroute measurements the R4 R5 link is expected to support the Santa Clara Richardson Santa Clara Tokyo and the Ottawa Tokyo traffic flows The other IP telephony traffic flows do not route over R4 R5 A peak of eight calls can be made over RA R5 for the four IP telephony ports per site 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 Enough link capacity The following table sorts the computations so that for each link the available link capacity is compared against the additional IP telephony load For example on link R4 R5 there is capacity 568 kbit s to allow for the additional 96 kbit s of IP telephony traffic Table 41 Link capacity example Link Utilization 96 Incremental IP telephony load Available End Capacity capacity Traffic Enough Points kbit s Threshold Used kbit s Site pair kbit s capacity R1 R2 1536 85 75 154 Santa Clara Ottawa 15 5 Yes Santa Clara Tokyo R1 R3 1536 R2 R3 1536 R2 R4 1536 R4 R5 1536 85 48 568 Santa Clara Richardson 24 Yes Ottawa Tokyo Santa Clara Tok
221. on Accept calls check box Box must be checked Routing Mode Direct Set to Direct Setup Q 931 not Nortel recommends that you always use Direct supported mode Call Control H 245 Check that call is active check box Leave box UNCHECKED every Enabling this feature will result in dropped calls 5 In the left frame click the Advanced button Ensure the following fields are set Table 32 Radvision Advanced screen required settings Field Value Description Check that the endpoint is check box Leave box checked online every ___ This setting controls the intervals when Radvision checks if the Business Communications Manager is still on line Enable TTL check box Box must be checked This is the only mechanism currently supported that allows the gatekeeper to determine if the end point the Business Communications Manager is active Force Direct for Service check box Check this box if you selected the Routing Mode Direct Calls on the Calls screen P0609327 02 Chapter 7 Optional VoIP trunk configurations 135 Gatekeeper support for interoperability 6 Create a service configuration for IPT a Select the Services tab b Click the Add button C Inthe Prefix field enter the unique telephone number that identifies the Meridian IPT system in the Business Communications Manager dialing plan 7 Define the IPT as a predefined endpoint a Select the Endpoints tab b Click the Add predefined button The Predefined
222. ones to your calculations As soon as an IP telephone is 5 c If you are using VoIP trunks do you need to activate MCDN features PSTN lines you do not need a separate MCDN keycode for VoIP trunks SIP trunks do not support the MCDN protocol Note If MCDN is already configured on your system for private networking over System configuration for IP functions Several sections of the Business Communications Manager must be properly configured prior to activation of IP telephony Answer the questions in the following table to determine if your Business Communications Manager has been correctly configured Table 6 Business Communications Manager system configuration Prerequisites Yes No 6 a Is the LAN functioning correctly with the Business Communications Manager 6 b Is the WAN functioning correctly with the Business Communications Manager 6 c Have you determined the published IP address for the system Refer to Finding the published IP address on page 35 6 d Have the necessary media gateway IP client and IP trunks resources been set Refer to Media gateway parameters for IP service 6 e Has a dialing plan been created taking into account special considerations for IP telephony and private and public networking P0609327 02 Chapter 2 Prerequisites checklist 35 Finding the published IP address The published IP address is the IP address used by comput
223. ons Manager is designed to interoperate with Radvision ECS 3 2 and CSE 1000 gatekeepers As part of this the Business Communications Manager supports both Direct GatekeeperResolved and Routed GatekeeperRouted call signaling in this mode of operation Note that if the call signaling method is changed the Business Communications Manager must be restarted before it functions properly Refer to Using a gatekeeper on page 133 for specific configuration instructions CSE 1000 gatekeeper note only supports GatekeeperResolved Network note Meridian 1 IPT systems do not support the Radvision gatekeeper NetCentrex gatekeeper BCM 3 6 and newer NetCentrex uses the GatekeeperResolvedNoRAS setting This requires some manual configuration such as entering the IP addresses of backup gateways and specifying some of the preferred characteristics Refer to Modifying local gateway settings for H 323 and SIP trunks on page 92 IP Telephony Configuration Guide 178 Interoperability Asymmetrical media channel negotiation Net Meeting By default the Business Communications Manager IP Telephony gateway supports the G 729 codec family G 723 1 G 711 mu law and G 711 A law audio media encoding Because NetMeeting does not support the H 323 fastStart call setup method NetMeeting can choose a different media type for its receive and transmit channels However Business Communications Manager IP Telephony gateway does not support calls with
224. option with each DHCP Discovery and Request message Additionally the IP telephone checks for either a vendor specific option message with a specific unique to Nortel IP telephones encapsulated sub type OR a site specific DHCP option In either case a Nortel IP telephone specific option must be returned by the IP telephone aware DHCP server in all Offer and Ack messages The IP telephone will use the information returned in this option to configure itself for proper operation This includes binding a new IP address netmask and gateway for local IP stack as well as configuring Server 1 minimum and optionally Server 2 By default Server 1 is always assumed to be the primary server after a DHCP session The IP telephone will not accept any Offers Acks if they do not contain e a Router option the IP telephone needs a default router to function AND e aSubnet Mask option AND e anSl Server Address and Port e The Nortel IP telephones require the scope value 128 to be configured on the DHCP server as follows Format example Nortel i2004 A iii jjj kkk lll ppppp aaa rrr iii jjj kkk 111 p ppp aaa rrr where Nortel i2004 A uniquely identifies this as the Nortel option Additionally the A signifies this version of this specification Future enhancements could use B for example ASCII is used to separate fields ASCII is used to separate Primary from Secondary Business Communications Manager information ASCII is used to signal e
225. or each enabled line All lines that are assigned to the same line pool should have the same programming 1 Click on the keys beside Services Telephony Services Lines VoIP lines Enabled VoIP lines Click on the General heading to enter a new name for the line a control set for the line Click on the Trunk line data heading and set the parameters you require for your system The line must belong to a line pool that contains the same type of VoIP line If you want specific restrictions assigned to the lines fill out the information under the Restrictions heading Repeat these steps for all the lines that are active Ensure that you put the H 323 trunks and SIP trunks in separate line pools Note Configuring SIP and H 323 trunks in the same line pool may result in unpredictable results since they do not support the same level of service SIP trunks for example do not support MCDN protocol services T 38 fax protocol or NetVision generated calls P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 109 Configuring telephones to access the VoIP lines For each telephone that will be allowed to use the VoIP line pools you must add the VoIP line pool to the DN record for that telephone 1 In Unified Manager open Services Telephony Services System DNs Active Set DNs DN XXX Line Access DN XXX is any DN that you want to allow to use VoIP trunking Click Line Pool Access Click Add The Add Lin
226. or the number of calls plus the overhead Determining WAN link resources For most installations IP telephony traffic travels over WAN links within the intranet WAN links are the highest recurring expenses in the network and they are often the source of capacity problems in the network WAN links require time to receive financial approval provision and upgrade especially inter LATA Local Access and Transport Area and international links For these reasons it is important to determine the state of WAN links in the intranet before installing IP telephony Link utilization This procedure explains how to determine and adjust link utilization 1 Get a current topology map and link utilization report of the intranet A visual inspection of the topology can indicate the WAN links anticipated to deliver IP telephony traffic 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 Determine the available spare capacity Business Communications Manager intranets are subject to capacity planning controls that ensure that capacity use remains below a determined utilization level For example a planning control can state that the utilization of a 56 kbit s link during the peak hour must not exceed 50 For a T1 link the threshold is higher at 85 The
227. ortel Networks 12050 Software Phone Configuring the 12050 Software Phone for the local system 1 Click the Start button and then click Settings 2 Click Control Panel 3 Double click the 12050 Software Phone icon The utility opens to the Communications Server tab as shown in the figure below Figure 15 i2050 Communications server i2050 Software Phone Properties E x Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound Devices Server Type C Obtain a server address automatically Use the following server address information PAddess 47 3 11 66 w Port puu C Name Cancel Apply Help IP Telephony Configuration Guide 72 Chapter3 Installing IP telephones 4 Enter the Published IP address of the Business Communications Manager in the IP address field 5 From the Port menu select BCM 6 Select the Server Type tab The screen shown in the following figure appears Figure 16 i2050 Switch type i2050 Software Phone Properties Hardware ID Advanced Audio Listener IP Trace About Communications Server Select Sound Devices Server Type C Meridian 1 C Centrex C CSE1000 C SL 100 C CSE6500 Enable Symposium Cancel Apply Help 7 Click the BCM option 8 Enable the Select Sound Devices tab for the USB headset To further configure this device through Unified Manager see Modifying IP telephone
228. otocols codec types and payload sizes Table 39 Bandwidth Requirements per Gateway port for half duplex links Ethernet B W PPP B W FR B W Payload Silence Silence Silence Codec Type Size NoSP Suppression NoSP Suppression NoSP Suppression ms peak peak Avg peak peak Avg peak peak Avg kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s kbit s G 711 30 163 2 163 23 163 23 150 4 150 43 150 43 154 2 154 23 154 23 64 kb s G 729 30 51 2 25 6 25 6 38 4 19 2 19 2 42 2 21 1 21 1 8 kb s G 723 1 30 48 0 24 0 24 0 35 2 17 6 17 6 39 0 19 5 19 5 6 3 kb s G 723 1 5 3 30 45 8 22 9 22 9 33 0 16 5 16 5 36 8 18 4 18 4 kb s Notes 1 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 IP Telephony Configuration Guide 156 Efficient Networking With no silence suppression both the transmit path and the receive path continuously transmit voice packets Therefore the peak bandwidth requirement per call on half duplex links is Peak Bandwidth per call 2 Continuous Transmission Rate Half Duplex links No Silence Suppression On half duplex links with silence suppression
229. ou want to assign to the handsets and you have all the line restrictions and telephony information you require to create or update a DN record for each telephone 4 Download the latest version of the NetVision Phone Administrator http www symbol com services downloads nvfirmware2 html Download the latest firmware version from the same website 5 You have obtained the Symbol NetVision serial cable whichis Purchased from Symbol at http used to transfer configuration information between the computer symbol com where the tool is installed and the handset part number 25 20528 01 6 You have a list of names that you will use for the handsets Each Name field name must be unique to a handset Both the H 323 Terminals record and the NVPA record must have exactly the same name IP Telephony Configuration Guide 76 Chapter4 Installing NetVision telephones 7 You have identified a PIN for each handset Password field 8 You have determined how you want to program codecs H 323 Terminals Record and General record Assigning general settings If you want your handsets to all use the same default codec and jitterbuffer use the settings on the General screen 1 Inthe Unified Manager click the keys beside Services IP Telephony and IP Terminals 2 Click H 323 Terminals 3 Click the General tab Figure 17 Defining Codec and Jitter Buffer for all terminals Summary Genera
230. ough DSP resources available The fallback feature can be in the Local Gateway Configuration With the fallback feature disabled calls move across the IP telephony trunks no matter what level of Quality of Service The fallback feature is active only at call setup A call in progress does not fall back if the quality degrades Calls fallback if there is no response from the destination an incorrectly configured remote gateway table or if there are not enough DSP resources available to handle the new call IP Telephony Configuration Guide 202 Quality of Service P0609327 02 203 Glossary access point 802 11b This is a piece of hardware using either IEEE 802 11 1 or 2 M bits sec Frequency Hopping Spread Spectrum or IEEE 802 11B 11 M bits sec Direct Sequence Spread Spectrum technology that connects to the internet and acts as a wireless gateway for devices to connect to the internet In the context of the Business Communications Manager this is the device that the NetVision handset uses to connect to the LAN to which the Business Communications Manager is connected backbone The major transmission path of a network handling high volume high density traffic bandwidth A measure of information carrying capacity available for a transmission medium shown in bits per second The greater the bandwidth the more information sent in a given amount of time bridge LAN equipment providing interconnection b
231. ough remote access packages and routing transferring the outside call to a VoIP trunk which is accessed by an allowed dial sequence The VoIP trunk connects directly to system B where the dialing sequence is recognized as directed to an internal DN In this scenario all remote call features are available to the caller IP Telephony Configuration Guide 110 Chapter6 Setting up VoIP trunks for outgoing calls Figure 32 Calling into a remote node from a public location Santa Clara Ottawa Target line XXX recognizes 2244 DN 3322 DN 2244 assigned with target line Xxx DN 2244 EE EE a a e T Dialin Eee en s S oos XXX 2044 00000 Gateway 3 A Remote gateway set Dialout up to Santa Clara CDP system code for Gateway destination digit 2 Ottawa 2 Route 022 VoIP DN type Private IP network Destination code 2 using route 022 dedicated VoIP trunk private network Absorb length 0 Ensure VoIP trunk is set up with remote filters Call process Based on the figure shown above this is how the call would progress 1 A home based employee in Santa Clara wants to call someone in Ottawa so they dial into the local Business Communications Manager network using the access code for an unsupervised trunk not VoIP trunks and the destination code and DN for the person they want to reach on System B Dialin XXX 2244
232. oy REA DA uk ad eae OS 127 Medic Aransas OOd ss 26 4 decide es awh bd 4 eH 8eRers sees deoewes 130 Media parameters record 00 seecc cee cee e non Rh vee ee RR aes 132 Radvision Calls screen required settings 2 00 eee eee eee 134 Radvision Advanced screen required settings 0 cee eee 134 Radvision Predefined Endpoints Properties settings 135 Cee 1000 FLSZS endpolilbs 4 eda card Rx soper gab CR bog Sob dabei x Reo dr 137 CSE 1000 H 323 dialing Plans sccciwedsccecemue doves km Rmo mech mes 138 CSE1000 codec compatibility with endpoints llle 139 CSE 1000 codec Configuration sas suu esum kx XR REG bm KRESS E 140 IP trunking interoperability fields soccer mmm 143 VoIP Transmission Characteristics for unidirectional continuous media stream 154 Bandwidth Requirements per Gateway port for half duplex links 155 IP Telephony Configuration Guide 14 Table 40 Table 41 Table 42 Table 43 Table 44 Table 45 Table 46 Table 47 Table 48 Table 49 Table 50 Table 51 Table 52 Table 53 Table 54 Bandwidth Requirements per Gateway port for Full duplex links 156 Link capacity BXAImpIg accncyccdcGaseneaceaseats soos sister nee eaans 161 Business Communications Manager 3 6 IP Interoperability Summary 175 Engineering SpDCIGBUO IS uaiuiuaca cede dex a a x on doppi XO eco qi 176 Supported voice payload sizes caw sa eewreekeaavarwacRen amen 176 Name COM
233. phony concepts on page 27 Business Communications Manager with voice over IP VoIP provides several critical advantages e Cost Savings IP networks can be significantly less expensive to operate and maintain than traditional networks The simplified network infrastructure of an Internet Telephony solution cuts costs by connecting IP telephones over your LAN and eliminates the need for dual cabling Internet Telephony can also eliminate toll charges on site to site calls by using your existing WAN By using the extra bandwidth on your WAN for IP Telephony you leverage the untapped capabilities of your data infrastructure to maximize the return on your current network investment e Cost flexibility The three models of IP telephones offer three levels of functionality that allow you to choose an IP telephone that fits your budget and or your service requirements e Portability and flexibility Employees can be more productive because they are no longer confined by geographic location IP telephones work anywhere on the network even over a remote connection With Nortel Networks wireless e mobility solutions your phone laptop or scanner can work anywhere on the network where a an 802 11b access point is installed Network deployments and reconfigurations are simplified and service can be extended to remote sites and home offices over cost effective IP links As well IP telephone functionality can be transferred between IP telephones using the
234. pplication from the Symbol web site and filled out the required information and determined what features will be added or deleted from the feature list Refer to the NetVision Phone Administrator Guide on your Business Communications Manager documentation CD or off the Symbol web site Do you have the necessary serial cable to perform the upload of handset information to the Business Communications Manager i series telephones Refer to Chapter 3 Installing IP telephones on page 41 NetVision wireless handsets Refer to Chapter 4 Installing NetVision telephones on page 73 P0609327 02 41 Chapter 3 Installing IP telephones An IP telephone converts the voice signal into data packets and sends these packets directly to another IP telephone or to the Business Communications Manager over the LAN or the internet e Ifthe destination is an IP telephone the arriving voice packets are converted to a voice stream and are routed to the speaker or headset of the target telephone e Ifthe destination is the Business Communications Manager the voice stream is routed to a circuit switched connection such as a telephone internal or line external PSTN or private network or some form of gateway VoIP Note IP telephones require an IP network to reach the Business Communications Manager However they do not need to use VoIP trunks to communicate beyond the Business Communications Manager They can use any
235. r this application Begin range This indicates the first port setting in the range 1024 65534 gt End lt range This indicates the last port setting in the range 1025 65535 gt 5 Click the Save button Port settings for legacy networks Business Communications Manager 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 one 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 Include port number UDP 5000 in the reserved port ranges for the QoS monitor e The port ranges reserved in a Business Communications Manager system are also reserved by the remote router You must reserve two ports for each voice call you expect to carry over the WAN link P0609327 02 Chapter 7 Optional VoIP trunk configurations 133 e You can reserve multiple discontinuous ranges Business Communications Manager requires that each range meet the following conditions Each range must start with an even number Eac
236. r configuration 57 Internet Control Message Protocol ICMP 173 Internet Engineering Task Force IETF 199 internet 3 way switch 46 Interoperability 175 interoperability gatekeeper supports 135 MWI on VoIP trunks 143 intranet delay and error analysis 162 networking multiple Business Communications Manager Systems 149 other resource considerations 162 routing changes 165 WAN link resources 153 Invalid Server Address 51 54 IP address DHCP configuration 53 gatekeeper 91 H 323 terminals list 81 network prerequisites 31 networking 36 private 36 122 public 36 122 Published IP address 35 remote gateway 103 IP address conflict 51 IP datagram 173 IP packet 154 IP speech packets 90 IP telephones 3 port switch 46 before installation 46 block single telephone 52 codec jitter buffer settings 57 codecs 45 57 viewing 51 P0609327 02 Index 215 contrast level 52 defined 22 deleting handset record 82 deregister 66 deregistering online sets 66 DHCP 33 display keys for configuration 48 does not connect 52 ethernet connection 47 feature labels 63 firmware downloading 65 H 323 Terminals record 79 home based network 152 12050 Software Phone 71 installing 41 73 Invalid server address 51 Jitter buffer 46 jitter buffer 57 Keep DN Alive 69 keycode 74 network check list 31 New telephone 51 No ports left 51 prerequisites 40 Published IP address 50 register prompt 51 registering 43 Registration disabled
237. ression Disabled The silence compression identifies periods of silence in a Enabled conversation and stops sending IP speech packets during those periods In a typical telephone conversation most of the conversation is half duplex meaning that one person is speaking while the other is listening If silence compression is enabled no voice packets are sent from the listener end This greatly reduces bandwidth requirements G 723 1 and G 729 support silence compression G 711 does not support silence compression Silence Compression Disabled v Enabled Disabled Performance note Silence Compression on all networked Business Communications Managers and IPT systems VAD setting on IPT systems must be consistent to ensure that interacting features such as Transfer and Conference work correctly As well the Payload size on the IPT must be set to 30ms P0609327 02 Chapter 5 Configuring local VoIP trunks 91 Table 22 Media parameters record Continued Field Value Description Jitter Buffer Voice Auto Select the size of jitter buffer you want to allow for your Small Jitter Buffer Voice AUTO Medium Large Refer to Jitter Buffer on page 28 T 38 Fax Support Enabled Note This field appears on H 323 screens only as SIP Disabled trunks do not support this feature Enabled The system supports T 38 fax over IP Disabled The system does not support T 38 fax over IP CAUT
238. rossing a slow WAN link IP Telephony Configuration Guide 200 Quality of Service Network Quality of Service This information under the headings in this section provides details about the quality of service aspects of networking e Network monitoring on page 200 e Quality of Service parameters on page 201 e Fallback to PSTN on page 201 Business Communications Manager VoIP Gateway uses a method like the ITU T Recommendation G 107 the E Model to determine the voice quality This model evaluates the end to end network transmission performance and outputs a scalar rating R for the network transmission quality The packet loss and latency of the end to end network determine R The model correlates the network objective measure R with the subjective QoS metric for voice quality MOS or the Mean Opinion Score This model provides an effective traffic building process by activating the Fallback to Circuit Switched Voice Facilities feature at call set up to avoid quality of service degradation New calls fall back when the configured MOS values for all codecs are below the threshold The model is the reason for compression 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 continuous basis The network monitoring fu
239. router to the telephone e S1 IP address 47 62 84 1 This is the published IP address of the Business Communications Manager The Business Communications Manager automatically assigns the telephone the DN of 3348 3 The installer configures DN record 3348 with the lines and attributes the IP telephone requires 4 The installer sets up a target line for DN 3348 using the Received Digits 3348 This phone would follow all of the same dialing rules as the other telephones on the Santa Clara Business Communications Manager A caller could dial 3321 to connect with telephone 3321 dial 9 to access the PSTN or dial 2 lt DN gt to access a telephone on the Ottawa system PSTN fallback metrics To view the metrics associated with VoIP calls that fall back to the PSTN network 1 Choose Diagnostics Service Metrics Telephony Services and click the PSTN fallback metrics heading The PSTN fallback metrics dialog shows metrics for e Last reset time e Fallback requests e Fallback failures Figure 41 Fallback Metrics fields PSTN fallback metrics Last reset time 20000101010000 Fallback requests o Fallback failures o P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 127 Resetting the log With PSTN Fallback metrics selected On the top menu click Configuration menu and select Clear data and time Quality of Service Monitor The Quality of Service Monitor is an application that monitors the qualit
240. rser is used for encoding and decoding oSIP from GNU software e SIP trunks are available between Business Communications Managers running BCM 3 5 or newer software T 38 fax restrictions and requirements Hardware restriction IP telephony including T 38 fax cannot operate on a system that has PEC Is installed on the MSC The following is a list of restrictions and requirements for the T 38 fax protocol Table 49 T 38 restrictions and requirements Supported Not supported only UDP transport MCDN only UDP redundancy TCP T 38 version 0 Forward Error Correction FEC on H 323 VoIP trunks between BCMs or between Fill removal BCMs and Meridian 1 IPT MMR transcoding JBIG transcoding Norstar systems SIP trunking Resource limitations T 38 fax transactions require significant DSP resources They use the same resources as the fax modem task Each task consumes one DSP or two DSPs if the session terminates on an application port such as voice mail Heavy fax traffic could affect IP telephone service if a number of faxes simultaneously come in on shared DSPs Refer to the Programming Operations Guide MSC section for details about setting up DSP configuration IP Telephony Configuration Guide 190 Interoperability P0609327 02 191 Appendix E Quality of Service The users of corporate voice and data services expect these services to meet a level of quality of service QoS This in turn affects network des
241. s 10 ms tftzrafl ca nortel com 10 10 10 1 2 1 ms 1 ms 1 ms 10 10 10 57 3 7 ms 2 ms 3 ms tcarrbf0 ca nortel com 10 10 10 2 4 8 ms 7 ms 5 ms bcarha56 ca nortel com 10 10 10 15 Trace complete The Traceroute program checks if routing in the intranet is symmetric for each source destination pairs Also the Traceroute program identifies the intranet links used to carry voice traffic For example if Traceroute of four site pairs gets the results shown in the following table you can calculate the load of voice traffic per link as shown in the second table Table 51 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 P0609327 02 Quality of Service 195 Table 52 Computed load of voice traffic per link Links Traffic from R1 R4 Santa Clara Richardson Santa Clara Tokyo R4 R5 Santa Clara Richardson Santa Clara Tokyo R5 R6 Santa Clara Richardson Richardson Ottawa R1 R2 Santa Clara Ottawa R5 R7 Santa Clara Tokyo R2 R3 Richardson Ottawa R3 R5 Richardson Ottawa Adjusting Ping measurements The Ping statistics are based on round trip measurements While the QoS metrics in the Transmission Rating model are one way To make the comparison compatible the delay and packet error Ping statistics are halved Refer to the information under the following headings for more details Adjustme
242. setup and how the Business Communications Manager fits into the network WAN A Wide Area Network WAN is a communications network that covers a wide geographic area such as state or country For Business Communications Manager 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 If you want to deploy IP telephones or NetVision telephones that will be connected to a LAN outside of the LAN that the Business Communications Manager is installed on you must ensure the Business Communications Manager has a WAN connection This includes ensuring that you obtain IP addresses and routing information that allows the remote telephones to find the Business Communications Manager and vice versa The Programming Operations Guide has a data section that describes the internet protocols and data settings that the Business Communications Manager requires or is compatible with Ensure that this connection is correctly set up and working before you attempt to deploy any remote IP devices LAN A Local Area Network LAN is a communications network that serves users within a confined geographical area For Business Communications Manager 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 A Business Communications
243. sfer your IP telephony configuration temporarily from one IP telephone to another using the Hot Desking feature This feature is described in detail in the Telephony Features Handbook You use FEATURE 999 to enter the feature To perform hot desking you are prompted for a password which is specified at the telephone before you can complete the task The Hot Desking password can be reset from the Unified Manager This allows users who forget their passwords to re enter hot desking and to reset their password Note This process also cancels hot desking for the telephone if the application is currently active Refer also to Notes about Hot Desking on page 62 To reset the Hot Desking password field for a specific IP telephone 1 2 3 Figure 11 Click the keys beside Services IP Telephony and IP Terminals keys Click Nortel IP Terminals Click the IP Terminal Status tab IP Terminal Status tab list Edit Configuration Performance Fault Report Tools Logoff View Help Summary General IP Terminal Status Telephony Features List IP Terminal Status 2431 Offline i2050 N Default N Default N 2432 Offline i2004 IN Default N A Default N A 2433 Offline i2002 IN Default N A Default N A 4 Select the IP telephone record you want to reset 5 On the top menu click Configuration then select Reset Hot Desking Password Performance Faull Modify parameters Deregister DN Forc
244. sing the Business Communications Manager with a Meridian 1 M1 IPT system or a Succession 1000 1000M system for Survivable Remote Gateway SRG applications there are also some interoperability settings to interact with these systems that need to be taken into consideration Refer to IP trunking interoperability settings on page 143 More VoIP trunk configuration e Setting up VoIP trunks for outgoing calls on page 101 provides information about setting up your VoIP trunks so your users can make calls to other systems e Optional VoIP trunk configurations on page 129 provides information about some applications or features that are not required for all trunks or which are optional to operation of the trunks Note VoIP trunks can be used for calls originating from any type of telephone within the Business Communications Manager system Calls coming into the system over VoIP trunks from other systems can be directed to any type of telephone within the system IP Telephony Configuration Guide 86 Chapter5 Configuring local VoIP trunks You cannot program DISA for voice over IP VoIP trunks therefore you cannot use COS passwords to remotely access features over your system The exception to this would be a tandemned call where a call comes into system A over the PSTN then tandems to system B over an VoIP trunk In this case the remote access package set up for the COS password will determine which system features are
245. ss Figure 3 Selecting the Published IP address sun Is NAT enabled Set the network interface with the most anticipated VoIP traffic as the Published IP address 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 Y Set the network interface on the public side as the published IP address Are all of your public side devices using a VPN Y Public Do you anticipate the most VOIP traffic on your public or private side Set the network interface on the public side as the published IP address Private Set the network interface on the private side as the published IP address The flowchart shown above makes reference to public and private IP addresses The public and private IP addresses are concepts relating to Network Address Translation NAT The decision also depends on whether a Virtual Private Network VPN is enabled For information about NAT and VPN refer to the Programming Operations Guide If you use IP telephones on the network they must be set to have the IP address of the network card they are connected to for their Default Gateway and the Published IP address as the S1 IP address For more information about this refer Configuring the 120XX telephones
246. ssion 3 0 X X Norstar IP Gateway X X IP Telephony Configuration Guide 184 Interoperability H 323 trunk compatibility issues The following tables provide a brief overview of the IP trunking and telephony compatibility issues including NetVision handset restrictions and Gatekeeper restrictions The tables are organized by Business Communications Manager software release numbers Table 47 Software interoperability restrictions and limitations for IP trunking Software release Description of restriction limitation All versions IPT payload sizes should be set to 30 ms All versions Silence suppression should be configured to the same value on both the Business Communications Manager and the M1 IPT for example both on or both off Silence suppression is called Voice Activity Detection on the M1 IPT 2 03 GA M1 IPT interaction with more than one IPT when transferring conferencing working with 25GA two or more IPT cards they must be on the same subnet If they are not on the same subnet one way speech path situations can occur 2 5 FP1 MR1 1 Gatekeeper e Officially Business Communications Manager supports only ECS 2 1 0 1 gatekeeper Business Communications Manager does not support Call Setup Q 931 routing mode e Business Communications Manager does not support the Radvision Dialing plan package e ECS option Check that call is active every XXX seconds must be unchecked e Radvision ECS
247. stration record 1 Inthe Unified Manager click the keys beside Services System DNs DN registration IP set DNs reg d 2 Click the key beside one of the following Active if you are deregistering an active IP telephone Inactive if you are deregistering an inactive IP telephone 9 DN Registration Active DNs reg d Inactive DNs reg d All DNs reg d DNs avail for reg n dictis reg d Registration type 5np4 Read Only Field DN 334 inactive 3 Select the DN for the IP Terminal you want to deregister IP Telephony Configuration Guide 68 Chapter3 Installing IP telephones 4 Click on the Configuration menu then select Deregister Refer to the figure below If you run Deregister on an active device you will be prompted to confirm that you understand that the device will be terminated If you click OK the device is deregistered immediately If you run Deregister on an inactive device there will be no prompts and the action will occur immediately Figure 14 Deregister DN from Configuration menu 9 IP set DNs reg d Q Active Registration type ian 4 nactive P0609327 02 Chapter 3 Installing IP telephones 69 Moving IP telephones IP telephones retain their DN when they are moved to a new location on the same subnet The following instructions apply to Nortel IP telephones Moving IP telephones and retaining the DN To move an IP telephone without
248. t determine voice quality for IP telephones and trunks e Codecs on page 27 e Jitter Buffer on page 28 e QoS routing on page 29 Codecs The algorithm used to compress and decompress voice is embedded in a software entity called a codec COde DECode Two popular Codecs are G 711 and G 729 The G 711 Codec samples voice at 64 kilobits per second kbps while G 729 samples at a far lower rate of 8 kbps For actual bandwidth requirements refer to Determining the bandwidth requirements on page 153 where you will note that the actual kbps requirements are slightly higher than label suggests Voice quality is better when using a G 711 CODEC but more network bandwidth is used to exchange the voice frames between the telephones If you experience poor voice quality and suspect it is due to heavy network traffic you can get better voice quality by configuring the IP telephone to use a G 729 CODEC Note You can only change the codec on a configured IP telephone if it is online to the Business Communications Manager or if Keep DN Alive is enabled for an offline telephone IP Telephony Configuration Guide 28 Chapter1 Introduction The Business Communications Manager supports these codecs e G 729 e G 723 e G 729 with VAD Voice Activity Detection e G 723 with VAD e GJll uLaw e GJll aLaw Jitter Buffer Voice frames are transmitted at a fixed rate because the time interval between frames is constant
249. te IP telephone site 152 Outgoing call configuration 102 107 outgoing calls 102 107 port ranges legacy systems 132 port settings 129 PSTN fallback 112 PSTN fallback schedule 118 Published IP address 35 QoS monitor status 127 remote access warning 109 110 remote gateway 103 106 routing 114 setting up target lines 98 signaling method 91 silence compression 90 target lines 98 trunk capacity 161 using a gatekeeper 133 using firewalls 129 VoIP trunks T 38 fax protocol 142 P0609327 02 Index 221 W WAN Business Communications Manager function 34 link resources 153 network engineering 158 Published IP address 35 Warning symbol 15 wireless IP 73 workstation prerequisites 40 Z zone ID MCDN 144 IP Telephony Configuration Guide 222 Index P0609327 02
250. telephones P0609327 02 Glossary 207 IP telephone In this book this term refers to any internet based telephone that works with the Business Communications Manager system For this release this includes the Nortel Networks IP telephones 12001 12002 12004 and 12050 Software Phone as well as the Symbol NetVision sets and NetVision data wireless handsets These telephones all interface to the Business Communications Manager LAN or WAN card through an internet or intranet link IPT This is the internet telephony gateway protocol for the Meridian 1 Business Communications Managers running BCM 3 5 or newer software require this protocol for trunk connections to the M1 The Business Communications Manager must be set to recognize that the other end of the trunk is an M1 IPT system Note IPT does not support the Radvision gatekeeper jitter buffer This is the process of collecting and organizing data frames at the receiving end to provide balanced voice quality kbit s kilobits per second Thousands of bits per second keycodes These are software codes that release feature applications on the Business Communications Manager such as VoIP trunks IP telephony ports and MCDN The Business Communications Manager Keycode Installation Guide provides generic instructions about obtaining keycodes and entering them into the Unified Manager latency The amount of time it takes for a discrete event to occur Mbit s M
251. telephones Model 7000 7100 7208 T7316 T7316E T7316E KIMs M7310 N M7324 N T7406 cordless telephone wireless telephones Companion DECT IP telephones and applications i series 200X and the Nortel Networks 12050 Software Phone and IP wireless telephones NetVision and NetVision Data telephones With this much flexibility the Business Communications Manager can provide the type of service you require to be most productive in your business While analog and digital telephones cannot be connected to the Business Communications Manager system with an IP connection they can make and receive calls to and from other systems through VoIP trunks Calls received through the VoIP trunks to system telephones are received through the LAN or WAN card and are translated within the Business Communications Manager to voice channels The IP telephones connect to the Business Communications Manager across an IP network through either a LAN or a WAN From the Business Communications Manager connection they can then use standard lines or VoIP trunks to communicate to other telephones on other public or private networks The Business Communications Manager also supports H 323 version 4 and H 323 third party devices through this type of connection Gatekeepers on the network A gatekeeper tracks IP addresses of specified devices and provides routing and optionally authorization for making and accepting calls for these devices A gatekeeper is not r
252. tems 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 Keycode Installation Guide For information about Resource Allocation see Configuring the MSC Resources in the Programming Operations Guide Each Business Communications Manager has 10 telephones that will be using VoIP lines In this setup only eight calls can be sent or received over the VoIP trunks at one time If all 10 telephones attempt to call at the same time two of the calls will be rerouted to the PSTN or other alternate routes if multiple routing is set up in the destination code schedule System programming for networking and fallback routes The following table provides the settings that are required for both systems to create a fallback network Table 27 Fallback configuration for to create fallback between two systems Task Settings for Santa Clara Settings for Ottawa Location in Unified Manager Set up a Control set for each 3321 2221 Services Telephony Services same line pool VoIP line Lines VoIP lines Enabled VoIP lines Set Published IP address that LAN 2 Services IP Telephony the devices on the Packet IP Terminals Data Network PDN will use to locate the system Set first preferred Codec G 729 Services IP Telep
253. ter reaches the interruptee the interruptee has spoken 2 jitter size frames past the intended point of interruption In cases where very large jitter sizes are used some users revert to saying OVER when they wish the other party to speak Possible jitter buffer settings and corresponding voice packet latency delay for the Business Communications Manager system IP telephones are e None e Small G 723 06 seconds G 711 G 729 05 seconds e Medium G 723 12 seconds G711 G 729 09 seconds P0609327 02 Chapter 1 Introduction 29 e Large G 723 18 seconds G711 G 729 15 seconds QoS routing To minimize voice jitter over low bandwidth connections the Business Communications Manager programming assigns specific DiffServ Marking in the IPv4 header of the data packets sent from IP telephones Warning BCM version 3 5 and newer software only supports H 323 version 4 To N support this all Business Communications Managers running BCM version 3 0 1 or earlier software which are on a network with a Business Communications Manager running BCM version 3 5 or newer software must either be upgraded to BCM version 3 5 or newer software or apply a QoS patch 3 0 0 25 or later to support this version of H 323 The DiffServ Code point DSCP is contained in the second byte of the IPv4 header DSCP is used by the router to determine how the packets will be separated for Per Hop Behavior PHB The DSCP is contained withi
254. the VoIP route is assigned to a schedule that can be activated from a control set For details about route and schedule configuration refer to the information under the headings below Adding routes for fallback on page 114 e Adding the destination code for the fallback route on page 116 e Setting up the VoIP schedule to overflow on page 118 e Activating the VoIP schedule for fallback on page 119 Pre configuration requirements e f you have not already done so remember to define a route for the local PSTN for your own system so users can still dial local PSTN numbers e Ensure the PSTN and VoIP line pools have been configured before you continue with this section For information about creating a VoIP line pool see Setting up the local gateway on page 91 Configure PSTN lines under Services Telephony Services Lines Physical Lines 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 Adding routes for fallback Enter the routes you want to use for normal and fallback traffic Add routes under Services Telephony Services Call Routing Routes Click the Add button to access the Add Routes dialog box Figure 34 Add route dialog box Add Routes Save Cancel P0609327 02 Chapter 6 Setting up VoIP trunks for outgoing calls 115 Add the PSTN route to other system
255. tings see Setting IP terminal general settings on page 43 Configuring telephone settings If you are not automatically registered to the Business Communications Manager you can configure your telephone settings to allow you to access a system on the network You will also need to perform these steps if your IP telephone is not connected to the same LAN to which the Business Communications Manager is connected Follow these steps to access the local configuration menu on an i200X telephone 1 Restart the telephone by disconnecting the power then reconnecting the power After about four seconds the top light flashes and NORTEL NETWORKS appears on the screen 2 When the greeting appears immediately and quickly press the four display buttons one at a time from left to right These buttons are located directly under the display Display buttons P0609327 02 Chapter 3 Installing IP telephones 49 Press the button sequence 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 f any other message appears disconnect then reconnect the power and try to access the configuration mode again Enter the network parameters as prompted As each parameter prompt appears use the keypad to define values Use the key to enter the period in the IP addresses Press OK to move forward The follow
256. tive database Configuration and activation information is described in the following sections Business Communications Manager requirements Set the Business Communications Manager Local Gateway IP interface to the following e Set Call Signaling Method to GatekeeperResolved e Set Gatekeeper IP to the IP address at which the CSE 1000 gatekeeper operates e Set Alias Names to a single H 323 identifier that is unique across all endpoints registered with the gatekeeper For example NAME BCM OTTAWA This H 323 identifier must exactly match that in the CSE 1000 gatekeeper configuration This entry is case sensitive Refer to the following sections for detailed information e CSE 1000 configuration adding an H 323 endpoint e Configuring Codec Compatibility P0609327 02 Chapter 7 Optional VoIP trunk configurations 137 CSE 1000 configuration adding an H 323 endpoint In the Gatekeeper Admin tool perform the following 1 Select GK standby DB admin 2 Select H 323 Endpoints 3 Select Add H 323 Endpoint 4 Ensure the following fields are set Table 34 CSE 1000 H 323 endpoints Field Value Description H323AliasName unique name This is the unique name that identifies your Business Communications Manager as an H 323 endpoint CDP Domain Name choose name from list If your system is using a CDP dialing plan choose the CDP domain name for the Business Communications Manager Tandem E
257. to IP telephony DHCP notes on page 54 Caution Do not enable DHCP on the Business Communications Manager if you have another DHCP server on the network Refer to the Programming Operations Guide for detailed information about disabling DHCP or about using other types of DHCP Setting up DHCP to work with IP terminals 1 Ensure that DHCP under Services is set up with the following settings Global Options tab NORTEL IP Terminal Information box is set to Nortel i2004 A lt ip address gt 7000 1 250 lt ip address gt 7000 1 1 Where lt ip address gt is the published IP address Be sure to include the period at the end of the string 1 250 Nortel IP Terminal VLAN ID contains an identification if the system is using the VLAN option If you do not know what the entry should be contact the system administrator for the VLAN switch If you want DHCP to automatically assign VLAN IDs to the IP telephones enter the VLAN IDs in the following format VLAN A id1 id2 idn Example if your VLAN IDS are 1100 1200 1300 and 1400 enter VLAN A 1100 1200 1300 1400 the entry must be terminated with a period If you do not want DHCP to automatically assign VLAN IDs to the IP telephones enter VLAN A none the entry must be terminated with a period Summary tab Status box is set to Enabled Ensure that the DHCP LAN settings are correct DHCP Local Scope LANX where LANX is a LAN that contains IP sets that use DHCP
258. to the system on page 47 P0609327 02 Chapter 2 Prerequisites checklist 37 Media gateway parameters for IP service To set up the media gateway resources that you require for optimum IP telephony and VoIP trunk service you need to define some basic gateway parameters These parameters are set in the System Configuration window Follow these steps to configure the media gateway 1 Click the Services and IP Telephony keys 2 Click System Configuration The Parameters screen appears in the right frame Figure 4 System Configuration Parameters screen 9 IP Telephony Parameters System Configuration Nortel IP Terminals Echo Cancellation Enabled w NLP H 323 Terminals H 323 Trunks G 723 1 Data Rate 6 3 kbps PortRanges Reserved Media Gateway Codec G 711 T 38 UDP Redundancy fo 3 Change the settings for the fields below as required for your system Table 8 IP terminals general record fields Field Value Description Echo Cancellation Enabled w NLP Enable or disable echo cancellation for your system Enabled Default Enabled w NLP check with your internet system Disabled administrator before changing this Echo Cancellation Enabled w NLP v G 723 1 Data Rate Reserved Media Gateway Codec Echo Cancellation selects what type of echo cancellation is used on calls that go through a Media Gateway NLP refers to Non Linear Processin
259. u want the telephone to have access to a Net6 content provider server have you also obtained the IP address for that server 8 b If DHCP is not being used has all telephone configuration been documented and made available for telephone installers Hint Use the Programming Record form Note If you are registering NetVision handsets to a system running DHCP ensure that you first enter a static IP list for all the handsets you intend to register 8 c If DHCP is not being used or if you want to enter the port manually has the VLAN port number been supplied if one is being used on the switch 8 d Have you determined the default codecs and payload sizes and default jitter buffers required by the IP network that supports the telephone 8 e Have telephone power and connectors been provisioned 8 f Do computers that will be using the Nortel Networks i2050 Software Phone meet the minimum system requirements including headset 8 g Do you want the system to auto assign DNs i series telephones If no complete 8 h Note If your company is using the Call Center application on the Business Communications Manager Nortel recommends that you manually assign DNs to avoid conflicts with Call Center DN assignments 8 h Have DN records been programmed for the corresponding IP clients use when manually assigning DNs to the telephones 8 1 NetVision handsets Have you obtained the current NetVision Phone Administrator a
260. unk configurations 131 Table 29 Media parameters record Continued Field Value Description Note You can reserve multiple discontinuous ranges Business Communications Manager requires that each range meet the following conditions Each range must start with an even number e Each range must end with an odd number e You cannot have a total of more than 256 ports reserved 4 Click the Save button The listing appears on the PortRanges screen Figure 43 Port Ranges PortRanges Modifying PortRanges Follow these steps to modify a port range 1 In Unified Manager open Services IP Telephony Port Ranges The PortRanges dialog box appears Refer to Figure 44 Figure 44 Port Ranges 9 IP Telephony PortRanges System Configuration Porth Ri Bed E ae R1 28000 28511 2 Select the Port Range you want to modify 3 From the top menu click Configuration and then select Modify PortRanges odify PortRanges M Add PortRanges Delete PortRanges The PortRanges dialog box appears Refer to Figure 42 IP Telephony Configuration Guide 132 Chapter 7 Optional VoIP trunk configurations Figure 45 Port ranges dialog box PortRanges Begin o End o Save Cancel 4 Enter the new port settings Table 30 Media parameters record Field Value Description PortRange R read only This field indicates the range of ports that are available fo
261. upported P0609327 02 Chapter 4 Installing NetVision telephones 75 Configuring NetVision records This section provides the steps for configuring the various records that the NetVision telephone requires to work on a Business Communications Manager system The information under the following headings describe e What information you require before you configure your handsets Gathering system information before you start e How to set up default codecs for all terminals Assigning general settings on page 76 How to determine the current status of H 323 on the system Monitoring H 323 service status on page 77 e How to set up an H 323 Terminals record on the Business Communications Manager to allow the NetVision handset to connect to the system Assigning H 323 Terminals records on page 78 Gathering system information before you start Ensure the following is complete or the information is on hand before you start configuring your NetVision telephones 1 If the system to which the handsets are registering is running DHCP ensure that you enter a static IP list for all the NetVision handsets you intend to create 2 The Business Communications Manager has been set up to Refer to Media gateway allow IP telephones parameters for IP service on page 37 3 If you are configuring the Business Communications Manager DN records records before you configure the handset You know which DNs y
262. ures on the NetVision handset P0609327 02 Chapter 4 Installing NetVision telephones 81 Modifying H 323 terminal records Once the handset registers to the system the H 323 terminal record appears on the H 323 Terminal List tab page From that entry you can modify or delete the record Updating the H 323 terminals record If you need to change the password for a NetVision telephone update the H 323 terminals record Follow these steps to update the H 323 Terminals record 1 Inthe Unified Manager click the Services IP Telephony IP Terminals keys 2 Click H 323 Terminals 3 Click the H 323 Terminal List tab 4 On the H 323 Terminal List screen highlight the terminal you want to change Summary General H 323 Terminal List H 323 Terminal List Name DN Passwod IPAddress Staus Codes PAT 470 1234 0 0 0 0 Out Of Service Default 5 At the top of the page click the Configuration menu and select Update Entry The H 323 Terminal List dialog box appears Figure 20 H 323 Terminal list with terminal information H 323 Terminal List Name PAT DN 470 Password 1234 IP Address Status Codec Default v Save Cancel 6 Enter a new password 7 Click the Save button IP Telephony Configuration Guide 82 Chapter4 Installing NetVision telephones Changing a handset Name The Name is the primary point of recognition for the Business Communications
263. user must enter on the handset to connect to the system from the handset You must enter at least four digits This is a mandatory field IP Address read only This field populates when the system assigns an IP address to the handset Status read only This field populates when the system registers the handset Codec Default Specifying a non default CODEC for a telephone allows you G 729 to override the general setting You might for example want G 711 uL to specify a low bandwidth CODEC like G 729 for a f aw telephone that connects to a remote or busy sub net G 711 aLaw If you choose Default the telephone will use the codec that is specified by the VoIP gateway it uses or what is determined by the gatekeeper if there is one 5 6 Click the Save button Note Shortly after the H 323 Terminals record is saved the system moves the DN you specified to the Active DNs list If you have not already done so configure the DN record for user requirements If you are not sure about how to configure DNs refer to the Programming Operations Guide for details about the various settings within this record Programming note Ensure that you choose Model IPWIs on the DN record General screen When the handset is registered check the handset feature menu and test the handset to ensure it is working as you expected Refer to the NetVision Telephone Feature User Card for directions about using Business Communications Manager call feat
264. using SWCA keys 7 Goto Services Telephony Services Lines Target Line Target line number from step 4 gt 8 Click the Trunk line data key 9 Click Received number 10 In the Public number field enter the DN The telephone assigned to that DN can now receive all calls with that DN number that come into the Business Communications Manager to which the telephone is connected For a detailed explanation about target lines see the Programming Operations Guide IP Telephony Configuration Guide 100 Chapter5 Configuring local VoIP trunks P0609327 02 101 Chapter 6 Setting up VoIP trunks for outgoing calls This section explains how to set up your system so that calls can be made from your Business Communications Manager system to other systems over VoIP trunks by identifying those systems to the Business Communications Manager Once the VoIP trunk is set up and the telephony programming is in place any type of telephone using your Business Communications Manager which has been assigned the VoIP line pool can use the trunk to call out of the system The following sections provide information about e Setting up remote gateways and end points on page 102 Configuration note If the VoIP network has a gatekeeper you do not need to configure remote gateways as the gatekeeper controls where the call packets go You do need to provide the gatekeeper administrator with your system settings so that calls are corr
265. ut from the DHCP server However a VLAN tag will be added to the packet at the switch port The packet will be untagged at the port of the IP phone P0609327 02 Interoperability 183 Symbol NetVision telephones In order to make calls between Symbol telephones and Business Communications Manager each must be configured to have at least one common codec The following codecs are supported by the NetVision telephones e G 711 u law e G711 A law e G 729 Annex A and Annex B Programming note If you are registering the handsets to a system which is using DHCP ensure that you first enter a static IP address for all the handsets you want to register Software interoperability compatibility and constraints The information under the following headings provides an overview of VoIP trunk compatibility issues e H 323 trunk compatibility issues e SIP trunk interoperability issues e T 38 fax restrictions and requirements H 323 trunk compatibility by software version The following table lists H 323 compatibility for each software version Table 46 Supported voice payload sizes BCM 2 5 Application FP1 MR1 BCM 3 0 BCM 3 0 1 BCM 3 5 BCM 3 6 H 323 v2 X X X H 323 v 4 X X ECS 2 1 0 1 X X ECS 3 0 X X X ITG 2 x 26 X X X IPT 3 0 X X T 38 patch X T 38 patch IPT 3 1 X X NetMeeting X X X X X Symbol DS X Symbol QCP X X CSE 1000 v 2 X X X X Succe
266. work use a non hub solution at its core P0609327 02 Chapter 2 Prerequisites checklist 33 Network assessment The following table questions are meant to ensure that the network is capable of handling IP telephony and that existing network services are not adversely affected Table 3 Network assessment Prerequisites Yes No 3 a Has a network assessment been completed been calculated 3 b Has the number of switch hub ports available and used in the LAN infrastructure 3 c Does the switch use VLANs If so get the VLAN port number and ID 3 d Have the used and available IP addresses for each LAN segment been calculated 3 e Has DHCP usage and location been recorded 3 f Has the speed and configuration of the LAN been calculated 3 g Has the estimated latency values between network locations been calculated 3 h Have the Bandwidth CIR utilization values for all WAN links been calculated 3 1 Has the quality of service availability on the network been calculated Resource assessment Answer the questions in the following table to determine if you have allocated sufficient resources on the Business Communications Manager for IP telephony For information about changing the DS30 split for the Business Communications Manager and allocating media resources refer to the Programming Operations Guide data sections Hardware restriction IP telepho
267. y bien y tu R i Hola Isabella Com o estas Tx Hello Fred This is Susan Do you have a minute Rx Th S Fre ere i Sure Conversation PA Bandwidth used nu Channel Link max Peak channel pandwicini is n average Bandwidth shared by half duplex calls bandwidth per ca IP Telephony Configuration Guide 170 Silence compression Silence compression on full duplex links On full duplex links the transmit path and the receive path are separate channels with bandwidths usually quoted in terms of individual channels The following figure shows the peak bandwidth requirements for one call on a full duplex link without silence compression Voice packets are transmitted even when a speaker is silent Therefore the peak bandwidth and the average bandwidth used equals the full transmission rate for both the transmit and the receive channel Figure 61 One call on a full duplex link without silence compression Tx Rx channel Bandwidth Tx channel Bandwidth Hello Fred This is Susan Do you have a minute Fred here Hi Sure Conversation 7 di Bandwidth used Channel Link max Channel Link N 77 Voice frames sent even when speaker is silent P0609327 02 Silence compression 171 When silence compression is enabled voice packets are only sent when a speaker is talking When a voice is being transmitted it uses the full rate transmiss
268. y for the Meridian 1 Ensure the dialing rules CDP or UDP are compatible with the M1 For information on CDP and UDP refer to the Programming Operations Guide Configure the PSTN fallback and enable QoS on both systems If target lines have not already been set up configure the telephones to receive incoming calls through target lines MCDN functionality on fallback PRI lines To be able to use MCDN functionality over PRI fallback lines e Check MCDN PRI settings on the M1 For information on this refer to the M1 documentation e Ensure SL 1 MCDN keycodes are entered on the Business Communications Manager and the PRI line is set up for SL 1 protocol For a detailed description of setting up fallback refer to Chapter 6 Setting up VoIP trunks for outgoing calls on page 112 P0609327 02 Chapter 8 Typical network applications using MCDN 149 Networking multiple Business Communications Managers You can also connect multiple offices with Business Communications Manager systems across your company Intranet This installation allows for CallPilot to direct calls throughout the system or for one system to support voice mail for the network Full toll bypass occurs through the tandem setup meaning that any user can call any DN without long distance charges being applied Users have full access to system users applications PSTN connections and Unified Messaging The network diagram shows two Business Communications Managers
269. y 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 indicated threshold it will trigger fallback to PSTN For information about setting up the system to use QoS and fallback to PSTN see Setting up VoIP trunks for fallback on page 112 Bandwidth required for QoS monitor There are monitoring packets that are sent back and forth between any two Business Communications Managers that are configured with each other as remote gateway entries to determine the available bandwidth for VoIP phone calls These packets are 88 bytes in length and are sent 100 times a minute at evenly spaced intervals in each direction The bandwidth required for this monitoring is then 2 X 100 X 88 bytes 60 seconds 293 bytes second or 2346 bits second in each direction for a total of 586 bytes second or 4693 bits second Warning Network note All systems in a private network must be running BCM 3 5 or A newer software or have the QoS 3 0 0 25 or later patch Business Communications Managers running BCM 3 0 1 or earlier software without installing the patch will be unable to support the new version of H 323 For further information about QoS refer to the information under the following headings e Quality of Service Status e Updating th
270. yo IP Telephony Configuration Guide 162 Efficient Networking Some network management systems have network planning modules that determine network flows These modules provide more detailed and accurate analysis because they can include correct node link and routing information They also help to determine network strength by conducting link and node failure analysis By simulating failures re loading network and re computed routes the modules indicate where the network can be out of capacity during failures Not enough link capacity If there is not enough link capacity consider one or more of the following options e Use the G 723 1 codec Compared to the default G 729 codec with 20 ms payload the G 723 1 codecs use 29 to 33 less bandwidth e Upgrade the bandwidth for the links Other intranet resource considerations Bottlenecks caused by non WAN resources do not occur often For a more complete evaluation consider the impact of incremental IP telephony traffic on routers and LAN resources in the intranet where the IP telephony traffic moves across LAN segments that are saturated or routers whose central processing unit CPU utilization is high Implementing the network LAN engineering To minimize the number of router hops between the systems connect the gateways to the intranet Ensure that there is enough bandwidth on the WAN links shorter routes Place the gateway and the LAN rout
271. ystem at the other end of the call must be set up to receive VoIP calls Outgoing call configuration details are explained under Managing H 323 and SIP trunks An example of an outgoing call over VoIP trunks is provided under PSTN call to remote node on page 109 Managing H 323 and SIP trunks The Business Communications Manager uses the same type of records for IP trunks that it creates records for physical lines and for target lines Found under Services Telephony Services Lines VoIP lines these records allow you to set some parameters about how the line will work When you have determined how you are going to split your trunks between H 323 and SIP trunks Counting IP trunks on page 87 you can configure the lines and put them into line pools which you use to create routing configurations Note that the H 323 lines start counting from the lowest position on the 60 line list and the SIP lines start from the top Once you have created the line pools you assign them to the telephones as you would any other line pool IP Telephony Configuration Guide 108 Chapter6 Setting up VoIP trunks for outgoing calls Keycodes 2 9 Lines VoIP lines Enabled VoIP lines Line 001 Line 002 H 323 trunks Line 003 Line 004 Line 057 Line 058 SIP trunks Line 058 Line 060 Configuring lines and creating line pools To set up the line configurations use the line record f

Avaya IP Telephony Configuration Guide

Contents

Download Pdf Manuals

Related Search

Related Contents