Part No. P0609327 02 March 17, 2004 Business Communications Manager 3.
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.
Contents Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 Before you begin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 Symbols used in this guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 Text conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents Chapter 3 Installing IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41 IP telephony on the Business Communications Manager . . . . . . . . . . . . . . . . . . . . . . 42 Configuring Nortel Networks i-series telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42 Preparing your system for IP telephone registration . . . . . . . . . . . . . . . . . . . . . . . 43 Setting IP terminal general settings . . . . . . . . . . . . . . . . . .
Contents Access points . . . . . . . . . . . . . . . . . Keycodes . . . . . . . . . . . . . . . . . . . . Handset and call functions . . . . . . . Codecs . . . . . . . . . . . . . . . . . . . . . . ...... ...... ...... ...... ...... ...... ...... ...... ....... ....... ....... ....... ...... ...... ...... ...... ....... ....... ....... ....... .... .... .... .... 5 73 74 74 74 Configuring NetVision records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents Managing H.323 and SIP trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107 Configuring lines and creating line pools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108 Configuring telephones to access the VoIP lines. . . . . . . . . . . . . . . . . . . . . . . . . 109 PSTN call to remote node . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109 Call process . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents 7 Setting Codecs on the CSE 1000 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139 Gatekeeper call scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141 Faxing over VoIP lines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142 Operational notes and restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142 IP trunking interoperability settings . . .
Contents Appendix B Silence compression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167 Silence compression on half-duplex links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168 Silence compression on full-duplex links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170 Comfort noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents 9 TCP traffic behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 199 Business Communications Manager router QoS support . . . . . . . . . . . . . . . . . . 199 Network Quality of Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200 Network monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200 Quality of Service parameters . . . . . . . . . . . .
Contents P0609327 02
Figures Figure 1 Network diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 Figure 2 Global IP settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35 Figure 3 Selecting the Published IP address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36 Figure 4 System Configuration, Parameters screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Figure 41 Fallback Metrics fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126 Figure 42 Port ranges dialog box . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130 Figure 43 Port Ranges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131 Figure 44 Port Ranges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Tables Table 1 Network diagram prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 Table 2 Network device checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32 Table 3 Network assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 Table 4 Resource assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 Table 5 Keycodes .
Table 40 Bandwidth Requirements per Gateway port for Full-duplex links . . . . . . . . . . . 156 Table 41 Link capacity example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161 Table 42 Business Communications Manager 3.6 IP Interoperability Summary . . . . . . 175 Table 43 Engineering specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 176 Table 44 Supported voice payload sizes . . . . . . . . . . . . . . . . .
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 i2050 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.
Preface Warning: Warning Symbol Alerts you to conditions where you can cause the system to fail or work improperly. Note: Note/Tip symbol Alerts you to important information. Tip: Note/Tip symbol Alerts you to additional information that can help you perform a task. ! Security Note: This symbol indicates a point of system security where a default should be changed, or where the administrator needs to make a decision about the level of security required for the system.
Preface 17 Acronyms This guide uses the following acronyms: API Application Programming Interface ATM Asynchronous Transfer Mode BCM Business Communications Manager CIR Committed Information Rate DID Direct Inward Dialing DOD Direct Outward Dialing DIBTS Digital In-Band Trunk Signaling DSB DIBTS Signaling Buffer DSL Digital Subscriber Line DSP Digital Signal Processor FEPS Functional Endpoint Proxy Server FoIP Fax over IP FUMP Functional Messaging Protocol ICMP Internet Control
Preface MB Mega Byte MOS Mean Opinion Score NAT Network Address Translation NVPA NetVision Phone Administrator PCM Pulse Code Modulation PING Packet InterNet Groper PiPP Power inline patch panel PPP Point-to-Point Protocol PRI Primary Rate Interface PSTN Public Switched Telephone Network QoS Quality of Service RAS Registration, Admissions and Status RTP Real-time Transfer Protocol SIP Session Initiation Protocol SNMP Simple Network Management Protocol TCP Transmission Con
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.
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 0800-810-500 Thailand 001-800-611-3007 Service Business Centre & Pre-Sales Help Desk +61-2-8870-5511 P0609327 02
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.
Chapter 1 • • Introduction 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. Increased customer satisfaction. Breakthrough e-business applications help deliver the top-flight customer service that leads to success.
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.
Chapter 1 Introduction Figure 1 Network diagram Business Communications Manager A Router LAN A PSTN Access Point IP telephone A SND MENU FCT RCL NAME 1 END 2 ABC 3 DEF 4 GHI 5 JKL 6 MNO 7 PQRS 8 TUV 9 WXYZ < 0 OPR # > CLR STO HOLD Digital telephone A NetVision (H 323 device A) I2050 telephone A Router WAN LAN B Gatekeeper Business Communications Manager B Inspe ct FORW ARD Calle rs MXP M1+IPT H 323 Device B IP telephone B Meridian set A Networking with Busin
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.
Chapter 1 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.
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).
Chapter 1 Introduction The Business Communications Manager supports these codecs: • • • • • • G.729 G.723 G.729 with VAD (Voice Activity Detection) G.723 with VAD G.711-uLaw G.711-aLaw Jitter Buffer Voice frames are transmitted at a fixed rate, because the time interval between frames is constant. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames.
Chapter 1 • Introduction 29 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 support this, all Business Communications Managers running BCM version 3.0.
Chapter 1 P0609327 02 Introduction
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.
Chapter 2 Prerequisites checklist Table 1 Network diagram prerequisites (Continued) 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.
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? 3.b Has the number of switch/hub ports available and used in the LAN infrastructure been calculated? 3.c Does the switch use VLANs? If so, get the VLAN port number and ID. 3.
Chapter 2 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 5.a Complete this question only if you are using VoIP trunks: Do you have enough VoIP keycodes? Both H.
Chapter 2 Prerequisites checklist 35 Finding the published IP address The published IP address is the IP address used by computers on the public network to find the Business Communications Manager. For example, if a Business Communications Manager has a LAN interface (LAN1) that is connected only to local office IP terminals and a WAN interface (WAN1) that is connected to the public network, then WAN1 should be set to the published IP address.
Chapter 2 Prerequisites checklist Determining the published IP address Use the flowchart in the following figure to determine which card should be set as the published IP address.
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.
Chapter 2 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 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.
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.b 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 between H.323 and SIP trunks. 7.
Chapter 2 Prerequisites checklist IP telephone records Complete this section if you are installing i-series and/or NetVision telephones. Table 10 IP telephone provisioning Prerequisites 8.a Are IP connections and IP addresses available for all IP telephones? If you want the telephone to have access to a Net6 content provider server, have you also obtained the IP address for that server? 8.
Chapter 3 Installing IP telephones An IP telephone converts the voice signal into data packets and sends these packets directly to another IP telephone or to the Business Communications Manager over the LAN or the internet. • • If the destination is an IP telephone, the arriving voice packets are converted to a voice stream and are routed to the speaker or headset of the target telephone.
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). • • The Nortel Networks i-series telephones use the UNISTIM protocol. The Symbol NetVision and NetVision Data telephones use H.323. Refer to Chapter 4, “Installing NetVision telephones,” on page 73.
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.
Chapter 3 3 Installing IP telephones 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 Off Set this value to ON to allow new IP clients to register with the system. WARNING: Remember to set Registration to Off when you have finished registering the new telephones.
Chapter 3 Installing IP telephones 45 Table 11 IP terminals general record fields (Continued) Field Value Description G.729 Payload Size (ms) 10, 20, 30, 40, 50, 60 Default: 30 G.723 Payload Size (ms) 30 Set the maximum required payload size, per codec, for the IP telephone calls sent over H.323 trunks. Note: Payload size can also be set for Nortel IP trunks. Refer to “Configuring media parameters” on page 89. G.
Chapter 3 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, 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.
Chapter 3 Installing IP telephones 47 The i2001, i2002, and newer models of the i2004 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 i200X telephone: 1 Connect one end of the handset cord to the handset jack on the telephone base. 2 Connect the other end of the handset cord to the handset.
Chapter 3 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: • • • • 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. 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.
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. • If any other message appears, disconnect, then reconnect the power, and try to access the configuration mode again. 3 Enter the network parameters, as prompted. As each parameter prompt appears, use the keypad to define values.
Chapter 3 Installing IP telephones Table 12 IP telephone server configurations (Continued) Field Value Description Full = 0 Partial = 1 If you indicate DHCP for the telephone, but you want to enter static IP addresses, choose 1 (Partial). If you choose 0 (Full), the DHCP server will assign IP addresses that are not static.
Chapter 3 Installing IP telephones 51 After you have entered all the configuration information, the telephone attempts to connect to the Business Communications Manager. The message Locating Server appears on the display. If the connection is successful, the message changes to Connecting to Server after about 15 seconds. Initialization may take several minutes. Do not disturb the telephone during this time.
Chapter 3 Installing IP telephones To see the Codec data for a telephone while it is on a call: Press the key, followed by the 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.
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.
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.
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.
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 In the Unified Manager, open Services, IP Telephony, IP Terminals and click Nortel IP Terminals. The IP Terminal summary screen appears.
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 In the 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.
Chapter 3 Installing IP telephones Figure 8 IP Terminal status dialog box 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 G.711-aLaw G.711-uLaw G.711 with VAD G.729 G.729 with VAD G.723 Specifying a non-default CODEC for a telephone allows you to override the general setting.
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.
Chapter 3 Installing IP telephones Figure 10 Add/Modify Telephony Features List 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 to use the features list.
Chapter 3 Installing IP telephones 61 Resetting the Hot Desking password You can transfer 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.
Chapter 3 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. The headset mode is not transferred by this feature.
Chapter 3 • Installing IP telephones 63 If the target telephone has a headset, 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.
Chapter 3 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
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).
Chapter 3 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.
Chapter 3 Installing IP telephones 67 Figure 13 Deregister DN from Configuration menu 5 Click Deregister DN. 6 Reregister the telephone, as described in “Configuring the i20XX telephones to the system” on page 47. 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 registration record: 1 In the Unified Manager, click the keys beside Services, System DNs, DN registration, IP set DNs reg’d.
Chapter 3 4 Installing IP telephones 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.
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 changing the DN: 1 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.
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. 1 At the telephone, enter FEATURE *510. 2 Press CHANGE. 3 Press * to toggle between + and - (minus), depending on which side of the time zone the telephone is located.
Chapter 3 Installing IP telephones 71 Nortel Networks i2050 Software Phone The Nortel Networks i2050 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 i2050 Software Phone uses the computer IP network connection to connect to the Business Communications Manager.
Chapter 3 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 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 status settings” on page 57.
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: • • • “NetVision connectivity” “Configuring NetVision records” on page 75 “Modifying H.
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.
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: • • • • What information you require before you configure your handsets (“Gathering system information before you start”). How to set up default codecs for all terminals (“Assigning general settings” on page 76).
Chapter 4 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 In the Unified Manager, click the keys beside Services, IP Telephony, and IP Terminals. 2 Click H.323 Terminals. 3 Click the General tab.
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 In the 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 3 The following table describes the fields on the screens. Table 19 H.
Chapter 4 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.
Chapter 4 Installing NetVision telephones 79 Adding a NetVision record in the Unified Manager Follow these steps to preconfigure an H.323 Terminals record for each handset you install: 1 In the Unified Manager, click the keys beside Services, IP Telephony and IP Terminals. 2 Click H.323 Terminals. 3 On the top menu, click Configuration, and choose Add Entry. The H.323 Terminal List dialog box appears. Figure 19 H.
Chapter 4 Installing NetVision telephones Table 20 H.323 Terminal list Field Value Description Name 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.
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.
Chapter 4 Installing NetVision telephones Changing a handset Name The Name is the primary point of recognition for the Business Communications Manager to identify a handset. If you need to change the name of an assigned handset: 1 Delete the existing record. Refer to “Deleting a NetVision telephone from the system” on page 82. 2 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.
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 NetVision 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.
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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.
Chapter 5 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 available to the caller.
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.
Chapter 5 4 Configuring local VoIP trunks 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.
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.
Chapter 5 4 Configuring local VoIP trunks 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 2nd Preferred Codec 3rd Preferred Codec 4th Preferred Codec None G.711-uLaw G.711-aLaw G.729 G.723 G.729 + VAD G.723 + VAD Select the Codecs in the order in which you want the system to attempt to use them.
Chapter 5 Configuring local VoIP trunks 91 Table 22 Media parameters record (Continued) Field Value Description Jitter Buffer - Voice Auto None Small Medium Large Select the size of jitter buffer you want to allow for your system. Refer to “Jitter Buffer” on page 28. T.38 Fax Support Enabled Disabled Note: This field appears on H.323 screens only, as SIP trunks do not support this feature. Enabled: The system supports T.38 fax over IP. Disabled: The system does not support T.
Chapter 5 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 In the 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.
Chapter 5 3 Configuring local VoIP trunks 93 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.
Chapter 5 Configuring local VoIP trunks Table 23 Local Gateway IP interface fields (Continued) Field Value Description *Gatekeeper IP If GateKeeperRouted, GateKeeperResolved or GateKeeperRoutedNoRAS are selected under Call Signaling, type the IP address of the machine that is running the gatekeeper. Backup gatekeeper(s) , NetCentrex gatekeeper does not support RAS, therefore, any backup gatekeepers must be entered in this field.
Chapter 5 Configuring local VoIP trunks 95 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. H245 Tunneling Disabled/Enabled Default: Disabled. If Enabled, the VoIP Gateway tunnels H.245 messages within H.225.
Chapter 5 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.
Chapter 5 Configuring local VoIP trunks 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 In the Unified Manager, click the keys beside Services, IP Telephony, and IP Trunking. 2 Click SIP Trunks.
Chapter 5 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, click the Services, IP Telephony, IP Trunks keys. 2 Click SIP Trunks. The Summary dialog appears.
Chapter 5 Configuring 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.
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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.
Chapter 6 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 MCDN Node C Ottawa VoIP trunk with MCDN Node 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.
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 VoIP call.
Chapter 6 Setting up VoIP trunks for outgoing calls Figure 30 Remote gateway dialog box 5 Use the information in the table below to set up the remote gateway information. Table 25 Remote gateway record Field Value Description Name Enter an indentifying tag for the remote system Destination IP Enter the IP address of the remote system gateway. QoS Monitor Disabled Enabled Choose Enabled, if you intend to use a fallback PSTN line.
Chapter 6 Setting up VoIP trunks for outgoing calls 105 Table 25 Remote gateway record (Continued) Field Value Description Receive 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 receive channel falls below this value for all of the available codecs, the BCM will fallback to a PSTN line.
Chapter 6 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).
Chapter 6 Setting up VoIP trunks for outgoing calls 107 Table 26 Adding SIP Address Book records (Continued) Field Value Description 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.
Chapter 6 Setting up VoIP trunks for outgoing calls Keycodes: 2 H.323 trunks SIP trunks Configuring lines and creating line pools To set up the line configurations, use the line record for 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.
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. 2 Click Line Pool Access. 3 Click Add. The Add Line Pool Access dialog box appears.
Chapter 6 Setting up VoIP trunks for outgoing calls Figure 32 Calling into a remote node from a public location Santa Clara Ottawa PSTN DN 3322 Target line XXX recognizes 2244 DN 2244 assigned with target line XXX DN 2244 Dialin: XXX-2244 Gateway: 2 Gateway: 3 Dialout: 2244 Gateway destination digit: 2 Route 022 (VoIP) DN type: Private Destination code: 2, using route 022 Absorb length: 0 Ensure VoIP trunk is set up with remote filters Remote gateway set up to Santa Clara CDP system code for
Chapter 6 Setting up VoIP trunks for outgoing calls Dialout: 2244 111 Ottawa Gateway: 3 dedicated VoIP trunk private network 4 System B recognizes the code as its own, and uses a local target line to route the call to the correct telephone.
Chapter 6 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).
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 IP network Business Communications Manager B Business Communications Manager A PSTN Public or Private PSTN line Public or Private PSTN line In a network configured for PSTN fallback, there are two connections between a Business Communications Manager and a remote system.
Chapter 6 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 the VoIP route is assigned to a schedule that can be activated from a control set.
Chapter 6 Setting up VoIP trunks for outgoing calls 115 Add the PSTN route to other system: 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 In the Route field, type a number between 001 and 999. This route defines the PSTN route to your local PSTN. 2 Click the Save button.
Chapter 6 Setting up VoIP trunks for outgoing calls 3 In the External # field: leave this field blank. 4 In the 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.
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.
Chapter 6 Setting up VoIP trunks for outgoing calls Normal. schedule for all fallback destination codes: Figure 37 Normal schedule routing information 1 Change Use Route to the route you configured for your PSTN fallback line (the line to the other system). 2 Set the Absorbed length to absorb the amount of the destination code that is not part of the DN for the other system.
Chapter 6 4 Setting up VoIP trunks for outgoing calls 119 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.
Chapter 6 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.
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 Both systems have destination code 9 set up as the local PSTN access code.
Chapter 6 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: • • • “System programming for networking and fallback routes” on page 123 “Making calls through a private VoIP network gateway” on page 125 “Connecting an i200X telephone” on page 125 In this scenario, shown in the following figure, two Business Communications Managers in different cities are connected th
Chapter 6 Setting up VoIP trunks for outgoing calls 123 • From this system, dial 9 to get onto PSTN • From this system, dial 9 to get onto PSTN • Dialing plan: CDP • Dialing plan: CDP, destination code is part of DN Routing Routing • Target DN 2244 (first digit is unique to system) • Target DN 3322 (first digit is unique to system) • Remote gateway destination digit: 2 • Remote gateway destination digit: 3 • Destination code: 2 • Destination code: 3 • VoIP/private network dialout:
Chapter 6 Setting up VoIP trunks for outgoing calls Table 27 Fallback configuration for to create fallback between two systems (Continued) Task Give all system telephones access to the VoIP line pool Settings for Santa Clara Settings for Ottawa 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
Chapter 6 Setting up VoIP trunks for outgoing calls 125 Table 27 Fallback configuration for to create fallback between two systems (Continued) Settings for Santa Clara Task Settings for Ottawa Location in Unified Manager Services, Telephony Services, Call routing, Routes, Route XXX Set up a route that contains the VoIP line pool. Route: 867 Dialout: N/A VoIP Line Pool: O DN type: Private Create a destination code that matches the Destination Digit(s).
Chapter 6 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: • • • Set IP address: 10.10.5.10 Default GW: 10.10.5.
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 quality of the IP channels. It does this by performing a check every 15 seconds. The QoS Monitor determines the quality of the intranet based on threshold tables for each codec.
Chapter 6 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.
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.
Chapter 7 Optional VoIP trunk configurations Follow these steps to add a port range: 1 In Unified Manager, open Services, IP Telephony, Port Ranges. 2 From the top menu, click Configuration, and then select Add PortRanges. The PortRanges dialog box appears. Refer to Figure 42. Figure 42 Port ranges dialog box 3 Enter the port settings.
Chapter 7 Optional VoIP trunk 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. • Each range must end with an odd number. • You cannot have a total of more than 256 ports reserved. 4 Click the Save button. The listing appears on the PortRanges screen.
Chapter 7 Optional VoIP trunk configurations Figure 45 Port ranges dialog box 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 for this application. Begin This indicates the first port setting in the range. End This indicates the last port setting in the range. 5 Click the Save button.
Chapter 7 • Optional VoIP trunk configurations 133 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. — Each range must end with an odd number. — You cannot have a total of more than 256 ports reserved. Using a gatekeeper This section describes the use of a gatekeeper for your H.323 VoIP trunks.
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.
Chapter 7 Optional VoIP trunk configurations 135 Gatekeeper support for interoperability: 6 7 Create a service configuration for IPT. a Select the Services tab. b Click the Add button. c In the Prefix field, enter the unique telephone number that identifies the Meridian IPT system in the Business Communications Manager dialing plan. Define the IPT as a predefined endpoint. a Select the Endpoints tab. b Click the Add predefined button. The Predefined Endpoint Properties dialog displays.
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:///gk/. Before an endpoint registers with the CSE 1000 gatekeeper it must first be added to the gatekeeper configuration.
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 This is the unique name that identifies your Business Communications Manager as an H.323 endpoint.
Chapter 7 6 Optional VoIP trunk configurations Ensure that the following fields are set: Table 35 CSE 1000 H.323 dialing plans Field Value Description Number This is the unique number that identifies the Business Communications Manager. Type This is the TON (Type of Number) or NPI (Numbering Plan Identifier) for the endpoint.
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.
Chapter 7 7 Optional VoIP trunk configurations Ensure the following fields are set: Table 37 CSE 1000 codec configuration Field Value Description Codec Name Name of the codec you selected. Voice Payload Size Choose the payload size for the codec. Use 30 ms for interoperability with the Business Communications Manager. Voice Playout (Jitter Buffer) Nominal Delay Choose the minimum jitter buffer value you want to allow.
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.
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.
Chapter 7 Optional VoIP trunk configurations 143 • 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. • 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.
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.
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 In the Tools menu, click Options. The options dialog box appears. Figure 48 NetMeeting options 3 Click Advanced Calling. The Advanced Calling Options dialog appears.
Chapter 7 Optional VoIP trunk configurations Figure 49 NetMeeting Advanced Calling Options 4 Under Gateway settings, select the Use a gateway option. 5 In the Gateway field, type the published IP address of the Business Communications Manager. 6 Click the OK button. 7 Add a remote gateway to your system as explained in “Setting up remote gateways and end points” on page 102. When prompted for the IP address of the remote gateway, type the IP address of the client computer.
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.
Chapter 8 Typical network applications using MCDN Figure 50 M1 to Business Communications Manager network diagram Head Office Warehouse M1 + IPT Business Communications Manager Meridian Telephone PSTN (fallback route) System telephone Intranet VoIP trunk Company server i2004 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 M1 ESN programming (CDP/UDP) is compatible.
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.
Chapter 8 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.
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 offices, each of which use a Business Communications Manager. A typical use of this would be a 1-800 number that users world-wide can call, who are then directed to the remote office best able to handle their needs.
Chapter 8 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 with call center” on page 151.
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.
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.
Efficient Networking 155 The peak bandwidth and average bandwidth requirements for a normal two-way call must take into account the affects of full and half duplex links and the affects of silence suppression. Refer to the tables in the next two sections, below, and to Table 40 on page 156 for voice Gateway bandwidth requirements. Peak bandwidth is the amount of bandwidth that the link must provide for each call.
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 enabled, the half-duplex nature of normal voice calls allows the sender and receiver to share the same bandwidth on the common channel.
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.
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 (DS0) 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.
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.
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.
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 Payload Size ms Codec Type G.
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.
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.
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: • • Improve meshing. Add links to help improve routing, adding a link from router1 to router4 instead of having the call routed from router1 to router2 to router3 to router4, reducing the hop count by two. Router reduction. Join co-located gateways on one larger and more powerful router.
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.
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Appendix B Silence compression This section describes using silence compression on half duplex and full duplex links: • • • “Silence compression on half-duplex links” on page 168 “Silence compression on full-duplex links” on page 170 “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.
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.
Silence compression 169 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.
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.
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 transmission rate. Since the sender and receiver do not share the same channel, the peak bandwidth requirement per channel is still equal to the full transmission rate. The following figure shows the peak bandwidth requirements for one call on a full-duplex link with silence compression enabled.
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.
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.
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Appendix D Interoperability This section discusses interoperability between the Business Communications Manager and other networks, including: • “Speech path setup methods” on page 176 • “Media path redirection” on page 177 • “Gatekeeper” on page 177 • “Asymmetrical media channel negotiation, Net Meeting” on page 178 • “Setting up Remote Routers for IP Telephony Prioritization” on page 179 • “Using VLAN on the network” on page 181 • “Symbol NetVision telephones” on page 183 • “Software i
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.
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: • 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.
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.
Interoperability 179 Setting up Remote Routers for IP Telephony 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).
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 Select a circuit in the Interfaces field. b Select a template in the Templates field. c Type a descriptive name in the Filter Name field. d Click the OK button. The Priority/Outbound Filters window opens. 11 Click the Apply button.
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 physical (or extended) LAN segment even though they may be geographically separated. VLAN IDs are determined by how the VLAN switch is configured.
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.
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. • G.711 u-law • G.711 A-law • 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.
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.
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 • 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. • 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.
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.
Interoperability 187 Table 47 Software interoperability restrictions and limitations for IP trunking (Continued) Software release Description 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.
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 version 2.03 BCM 2.5* 2.5 FP1* 2.5 FP1 BCM 3.0/ BCM Net MR1* 3.0.1* 3.5 Meeting BCM 2.03 X BCM 2.5 X X BCM 2.5 FP1 X X X X X X X X X FP1 MR 1.1 BCM 3.0 ITG/IPT v. X.X Symbol GK CSE1K basic call to/from ITG v. 25.
Interoperability • • • • 189 SIP trunks do not support the MCDN networking protocol Business Communications Manager call redirection and conferencing are supported a third-party SIP parser is used for encoding and decoding -- oSIP from GNU software 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.
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Appendix E Quality of Service The users of corporate voice and data services expect these services to meet a level of quality of service (QoS). This, in turn, affects network design. The purpose of planning is to design and allocate enough resources in the network to meet user needs. QoS metrics or parameters help in meeting the needs required by the user of the service.
Quality of Service Figure 64 Relationship between users and services Delay variation Business Communications Manager IP telephony parameters - Fallback threshold - Codec - Silence compression - Echo cancellation - Non-linear programming Business Communications Manager VoIP Corporate intranet Deliver voice/fax service Deliver IP service User oriented QoS - Roundtrip conversation delay - Clipping and dropout - echo Network QoS metrics - One-way delay - Packet loss The IP gateway can monitor the
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.
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.
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.
Quality of Service Measurement procedure The following procedure is an example of how to get delay and error statistics for a specific site pair during peak hours. Program a script to run the Ping program during the intranet peak hours, repeatedly sending a series of 50 Ping requests. Each Ping request generates a summary of packet loss, with a granularity of 2%, and, for each successful probe that made its round-trip, that many rtt samples.
Quality of Service 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.
Quality of Service Implementing QoS in IP networks The information under the headings in this section explain how to implement QoS in IP networks: • • “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.
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 occurs the throughput decreases, and the whole cycle repeats.
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. • • • “Network monitoring” on page 200 “Quality of Service parameters” on page 201 “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.
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. • • • Packet loss: Packet loss is the percentage of packets that do not arrive at their destination.
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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.
Glossary communications protocol A set of agreed-upon communications formats and procedures between devices on a data communication network. data communications Processes and equipment used to transport signals from a data processing device at one location to a data processing device at another location. default gateway For IP telephony, this refers to the router that closest to the IP telephone.
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.
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 Manager IP Telephony supports H.323.
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 (i2001, i2002, i2004 and i2050 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.
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.
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.
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.
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 Services, Radvision 135 assessment network 33 resources, prerequisite 33 asymmetrical media channel negotiation 178 routing 165 Asynchronous Transfer Mode (ATM) 154 B background noise 172 bandwidth available for othe
Index remote gateway 103 comfort noise 172 configure DN record 51 i2050 Software Phone 71 IP server parameters 49 restart to 48 review information 51 DHCP configuring 53 configuring for IP telephones 53 Invalid Server Address 54 IP telephone 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 Connecting to Server 51 dialed digits, VoIP trunk routing 114 contrast level, IP telephones 52 CSE MCDN fo
Index echo reply 173 Frame Relay 154 efficient networking 153 full duplex link bandwidth requirements 156 silence compression examples 170 silence suppression 157 VoIP load 161 WAN engineering 158 Enable TTL 134 end to end delay 163, 193 end to end DTMF signaling 176 Endpoint Type, Radvision 135 213 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 s
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.
Index contrast level 52 defined 22 deleting handset record 82 deregister 66 deregistering online sets 66 DHCP 53 display keys for configuration 48 does not connect 52 ethernet connection 47 feature labels 63 firmware, downloading 65 H.
Index Keep DN alive 69 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 mu-law 178 multi-locations, networking 151 MWI VoIP trunk interoperability 143 N name changing on handset 82 H.323 terminals list 81 H.
Index 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 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 traf
Index using M1-IPT 25 queuing delay 163 PRI, MCDN fallback 148 private IP address 32, 36, 122 R private network, MCDN Zone ID 144 R1 determining link capacity 161 peak VoIP load 161 private network, virtual ID 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 12
Index routes full duplex link 161 recording 194 site pairs 194 full duplex links 157 half duplex links 156 SIP fallback setting 93 routing and hop count 163 asymmetrical 165 delay issues 165 instability 165 network example 125 PSTN fallback 118 VoIP trunks 114 site pairs 194 S slow connection, IP telephones 52 S1 Action 50 S1 IP 50 S1 Port 50 S1 RETRY Count 50 S2 Action 50 S2 IP 50 S2 Port 50 S2 RETRY Count 50 schedule activating VoIP schedule 119 control set 125 destination codes 117 PSTN fallback 11
Index traffic network loading 160 network mix 198 WAN link resources 153 voice path, silence suppression 156 transfer media path redirection 177 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 transmission characteristics 154 transmit fallback threshold 192 transmit path 156 Transmit Threshold 103, 124 troubleshooting IP telephones 51 network delay
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
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