CHAPTER 3 Use the information in this chapter to prepare for the installation of the Cisco Unified MeetingPlace system telephony and LAN components. This chapter contains the following sections: Cisco Unified MeetingPlace Components, page 3-1 Selecting a Site, page 3-1 Worksheets, page 3-2 Cisco Unified MeetingPlace Components The Cisco Unified MeetingPlace system includes the following hardware and software components: The Audio Server hardware is a call- and voice-processing hardware platform that connects to the phone network and to a compatible LAN. The Audio Server software controls the platform and provides Cisco Unified MeetingPlace functions to desktops on the LAN. It also provides digital telephony access to PSTN callers and IP telephony access to Voice over IP (VoIP) callers. The Audio Server software communicates with other Cisco Unified MeetingPlace components over the LAN or WAN. (For details, see the About Cisco Unified MeetingPlace Integration Applications section on page 1-21.) Selecting a Site The Cisco Unified MeetingPlace 8100 series hardware is usually installed in an equipment room (for example, a PBX or computer room). The location must meet the Cisco Unified MeetingPlace system environmental and power requirements and must allow you to connect the system to the phone network and LAN. For environmental requirements and hardware specifications for the Cisco Unified MeetingPlace 8100 series server, refer to the Preparing to Install the Cisco Unified MeetingPlace 8100 Series Server chapter of the Installation and Upgrade Guide for Cisco Unified MeetingPlace Audio Server http://www.cisco.com/en/us/products/sw/ps5664/ps5669/prod_installation_guides_list.html. This section contains the following information: LAN Workstation Minimum Configuration, page 3-2 Sensitivity to Network Traffic, page 3-2 3-1
Worksheets Chapter 3 Caution This equipment has been tested and found to comply with the limits for a Class A digital device pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case the user will be required to correct the interference at his own expense. LAN Workstation Minimum Configuration You must install additional MeetingTime software on Microsoft Windows software-based desktops for use by Cisco Unified MeetingPlace end users, contacts, and attendants. Sensitivity to Network Traffic Because Cisco Unified MeetingPlace is connected to the LAN network, traffic on the local LAN segment can affect Cisco Unified MeetingPlace operations. In particular, a broadcast storm (one or more systems on a network segment continuously sends message packets to the local broadcast address) can bring a Cisco Unified MeetingPlace system down for the duration of the storm. Unless the system administrator is certain that broadcast storms will not occur, the Cisco Unified MeetingPlace 8106 or 8112 server should be partially isolated from the rest of the network. To isolate the server, you can use an Ethernet router or switch. An Ethernet switch is usually simpler and less expensive than a router. Many switches include filtering mechanisms that control broadcasts. (For example, the 3COM LinkSwitch 1000 temporarily shuts down any port that generates an abnormal number of broadcast messages.) A Cisco Unified MeetingPlace system connected directly to a filtering switch is virtually immune to broadcast storms. A Cisco Unified MeetingPlace server placed on its own router segment is also immune to broadcast storms. A moderately expensive solution is to place a two-ethernet-port router between the Cisco Unified MeetingPlace system and the main network. An example of such a device is the Cisco 2514. Caution A router-based solution implies creating a new LAN segment with its own IP address range. Worksheets Use the worksheets below to gather information for the telephony- and LAN-planning portion of your Cisco Unified MeetingPlace system installation. Each worksheet includes a list of requirements that you must provide to ensure a successful installation. Worksheet 3-1: Cisco Unified MeetingPlace Telephony Requirements for Non-ISDN T1 Lines (U.S., Canada, Japan), page 3-4 Worksheet 3-2: Cisco Unified MeetingPlace Telephony Requirements for ISDN-PRI Lines (U.S., Canada, Japan), page 3-6 Worksheet 3-3: Cisco Unified MeetingPlace Telephony Requirements (Hong Kong), page 3-8 Worksheet 3-4: Cisco Unified MeetingPlace Telephony Requirements (Europe), page 3-10 3-2
Chapter 3 Worksheets Worksheet 3-5: Cisco Unified MeetingPlace LAN Requirements (Worldwide), page 3-12 Worksheet 3-6: Cisco Unified MeetingPlace VoIP Phone Connections Requirements (Worldwide), page 3-14
Chapter 3 Worksheet 3-1: Cisco Unified MeetingPlace Telephony Requirements for Non-ISDN T1 Lines (U.S., Canada, Japan) Worksheet 3-1: Cisco Unified MeetingPlace Telephony Requirements for Non-ISDN T1 Lines (U.S., Canada, Japan) This worksheet describes the telephony requirements for non-isdn T1 lines for the United States, Canada, and Japan. 1. Access ports Number of access ports (user licenses) your company purchased: # 2. Telephony setup How Cisco Unified MeetingPlace will attach to the phone network: Directly to PSTN (U.S. and Canada only) Through a PBX 3. T1 lines Order T1 lines to accommodate the number of T1 ports purchased. 4. Trunking type The trunking type provided to the Cisco Unified MeetingPlace system: Digital trunks 5. PBX requirements Order additional hardware or software for your PBX to accommodate new T1 lines. The following telephony components or services can take from 4 to 6 weeks to order and install: 6. Cisco Unified MeetingPlace phone number(s) Network Telco Service (trunk lines, main phone number, or combined access numbers) PBX specific hardware or software (Smart Blades, software upgrade) Additional hardware (CSU, UPS) 7. Digital line requirements Standard or fractional T1 The main phone number (800 and/or local) at the beginning of the range. Channelized into 24 channels Ability to receive calls Ability to place calls (required for outdials and alarm outcalls) Ability to hunt all ports (linear, circular, UCD, or ACD) Toll restriction on ports lifted (required for outdials and alarm outcalls) DTMF generated by all PBX or network phones, including the operator console Male RJ-48C jack connector (U.S. and Canada only) Note PBX/Telco must provide disconnect supervision. If using DID Meeting Access, the numbering plan for DID/DNIS or DDI, including ranges for Combined, Profile, and Direct Meeting Access. 3-4
Chapter 3 Worksheet 3-1: Cisco Unified MeetingPlace Telephony Requirements for Non-ISDN T1 Lines (U.S., Canada, Japan) Signaling E&M wink Start (default configuration) Note MeetingTime supports only line-side loop-start and Ground Start CAS protocols but supports both line-side and trunk-side Wink Start CAS protocols. Ground Start from local Telco service E&M Wink Start, DTMF digits, and DID/DNIS or DDI Clear Channel Loop start (OPS) Coding format B8ZS (strongly recommended) or one of the following: Jammed Bit (bit stuffing) AMI Note If AMI is used, the number of zeros (0) in a row may exceed the recommended Telco value of 15. This can happen if all participants in the same meeting are on the same span. In that case, a 0s pattern could be sent to all channels at the same time, exceeding the maximum. This, in turn, can lead to meeting participants being dropped by the Telco from the meeting. Therefore, use B8ZS or Jammed Bit, if possible. Framing One of the following: Extended super frame (ESF) (strongly recommended) Standard super frame D4 (SF/D4) Note T1 facilities using D4 framing are susceptible to false triggering of yellow alarm signals. This false triggering can lead to dropped calls. Conferencing applications are particularly susceptible to this problem with D4 framing. The problem can occur when all 24 channels on a single span are in a conference and are carrying identical data streams. D4 transmission equipment in the network can falsely interpret the identical data on all 24 channels as a yellow alarm signal. For this reason, we recommend configuring T1 spans for ESF framing. Additional hardware Channel service unit (CSU) required for each T1 connection over 330 feet (100 meters) from the demarcation point (demarc) or from the PBX. 8. Modem requirements Analog phone line, or PBX connection Pilot number, accessible from outside RJ-11C connector 9. Translation table Translation table requirements for dialing restrictions. 3-5
Worksheet 3-2: Cisco Unified MeetingPlace Telephony Requirements for ISDN-PRI Lines (U.S., Canada, Japan) Chapter 3 Worksheet 3-2: Cisco Unified MeetingPlace Telephony Requirements for ISDN-PRI Lines (U.S., Canada, Japan) This worksheet describes the telephony requirements for ISDN-PRI lines for the United States, Canada, and Japan. 1. Access ports Number of access ports (user licenses) your company purchased: # 2. Telephony setup How Cisco Unified MeetingPlace will attach to the phone network: Directly to PSTN (U.S. and Canada only) Through a PBX 3. ISDN-PRI lines Order ISDN-PRI lines to accommodate the number of ISDN-PRI ports purchased. 4. Trunking type The trunking type provided to the Cisco Unified MeetingPlace system: Digital trunks 5. PBX requirements Order additional hardware or software for your PBX to accommodate new T1 lines. The following telephony components or services can take from 4 to 6 weeks to order and install: 6. Cisco Unified MeetingPlace phone number(s) Network Telco Service (trunk lines, main phone number, or combined access numbers) PBX specific hardware or software (Smart Blades, software upgrade) Additional hardware (CSU, UPS) 7. Digital line requirements Standard or fractional T1 The main phone number (800 or local) at the beginning of the range. Channelized into 24 channels (with 23 B-channels and one D-channel in timeslot 24) NFAS (Non-Facility Associated Signaling with one ISDN trunk controlled by the D-channel of another) is not allowed Ability to receive calls Ability to place calls (required for outdials and alarm outcalls) Ability to hunt all ports (linear, circular, UCD, or ACD) Toll restriction on ports lifted (required for outdials and alarm outcalls) DTMF generated by all PBX or network phones, including the operator console Male RJ-48C jack connector (U.S. and Canada only) Note PBX/Telco must provide disconnect supervision. If using DID Meeting Access, numbering plan for DID/DNIS or DDI, including ranges for Combined, Profile, and Direct Meeting Access 3-6
Chapter 3 Worksheet 3-2: Cisco Unified MeetingPlace Telephony Requirements for ISDN-PRI Lines (U.S., Canada, Japan) Signaling AT&T TR41459 ISDN (default configuration) Nortel DMS-100 ISDN Telcordia Technologies NI-2 ISDN Coding format B8ZS (strongly recommended), or one of the following: Jammed Bit (bit stuffing) AMI Note If AMI is used, the number of zeros (0) in a row may exceed the recommended Telco value of 15. This can happen if all participants in the same meeting are on the same span. In that case, a 0s pattern can be sent to all channels at the same time, exceeding the maximum. This, in turn, can lead to meeting participants being dropped by the Telco from the meeting. Therefore, use B8ZS or Jammed Bit, if possible. Framing One of the following: Extended super frame (ESF) (strongly recommended) Standard super frame D4 (SF/D4) Note T1 facilities using D4 framing are susceptible to false triggering of yellow alarm signals. This false triggering can lead to dropped calls. Conferencing applications are particularly susceptible to this problem with D4 framing. The problem can occur when all 24 channels on a single span are in a conference and are carrying identical data streams. D4 transmission equipment in the network can falsely interpret the identical data on all 24 channels as a yellow alarm signal. For this reason, Cisco recommends configuring T1 spans for ESF framing. Additional hardware Channel service unit (CSU) required for each T1 connection over 330 feet (100 meters) from the demarcation point (demarc) or from the PBX. 8. Modem requirements Analog phone line, or PBX connection Pilot number, accessible from outside RJ-11C connector 9. Translation table Translation table requirements for dialing restrictions. 3-7
Worksheet 3-3: Cisco Unified MeetingPlace Telephony Requirements (Hong Kong) Chapter 3 Worksheet 3-3: Cisco Unified MeetingPlace Telephony Requirements (Hong Kong) This worksheet describes the non-isdn T1 CAS telephony requirements for Hong Kong. Note If you are planning a T1-PRI telephony installation for Hong Kong, use the information in Worksheet 3-2: Cisco Unified MeetingPlace Telephony Requirements for ISDN-PRI Lines (U.S., Canada, Japan), page 3-6. 1. Access ports Number of access ports (user licenses) your company purchased: # 2. Telephony setup How Cisco Unified MeetingPlace will attach to the phone network: Directly to PSTN Through a PBX 3. T1 lines Order T1 lines to accommodate the number of T1 ports purchased. 4. Trunking type The trunking type provided with the Cisco Unified MeetingPlace system: Digital trunks 1.544 Mbps T1 IDA M service 5. PBX requirements Order additional hardware or software for your PBX to accommodate new T1 lines. The following telephony components or services can take from 4 to 6 weeks to order and install: 6. Cisco Unified MeetingPlace phone number(s) Network Telco Service (trunk lines, main phone number, or combined access numbers) PBX specific hardware or software (Smart Blades, software upgrade) Additional hardware (CSU, UPS) 7. Digital line requirements Standard or fractional T1 The main phone number (800 and/or local) at the beginning of the range. Channelized into 24 channels Ability to receive calls Ability to place calls (required for outdials and alarm outcalls) Ability to hunt all ports (linear, circular, UCD, or ACD) Toll restriction on ports lifted (required for outdials and alarm outcalls) DTMF generated by all PBX or network phones, including the operator console Male RJ-45 jack connector Note PBX/Telco must provide disconnect supervision. 3-8
Chapter 3 Worksheet 3-3: Cisco Unified MeetingPlace Telephony Requirements (Hong Kong) Signaling One of the following: If using DID Meeting Access, numbering plan for DID/DNIS or DDI, including ranges for Combined, Profile, and Direct Meeting Access E&M Wink Start E&M Wink Start, DTMF digits, and DID/DNIS or DDI Loop start only Coding format One of the following: B8ZS (preferred) Jammed Bit (bit stuffing) AMI Note If AMI is used, the number of zeros (0) in a row may exceed the recommended Telco value of 15. This can happen if all participants in the same meeting are on the same span. In that case, a 0s pattern can be sent to all channels at the same time, exceeding the maximum. This, in turn, can lead to meeting participants being dropped by the Telco from the meeting. Therefore, use B8ZS or Jammed Bit, if possible. Framing One of the following: Extended super frame (ESF) (preferred) Standard super frame D4 (SF/D4) Note Triggering of yellow alarm signals. This false triggering can lead to dropped calls. Conferencing applications are particularly susceptible to this problem with D4 framing. The problem can occur when all 24 channels on a single span are in a conference and are carrying identical data streams. D4 transmission equipment in the network can falsely interpret the identical data on all 24 channels as a yellow alarm signal. For this reason, Cisco recommends configuring T1 spans for ESF framing. Additional hardware Channel service unit (CSU) required for each T1 connection over 330 feet (100 meters) from the demarcation point (demarc) or from the PBX. 8. Modem requirements Analog phone line, or PBX connection Pilot number, accessible from outside RJ-11C connector 9. Translation table Translation table requirements for dialing restrictions. 3-9
Worksheet 3-4: Cisco Unified MeetingPlace Telephony Requirements (Europe) Chapter 3 Worksheet 3-4: Cisco Unified MeetingPlace Telephony Requirements (Europe) This worksheet describes the telephony requirements for Europe. 1. Access ports Number of access ports (user licenses) your company purchased: # 2. Telephony setup How Cisco Unified MeetingPlace will attach to the phone network: Directly to PSTN Through a PBX 3. E1 lines Order E1 lines to accommodate the number of E1 ports purchased. 4. Trunking type The trunking type provided with the Cisco Unified MeetingPlace system: Digital trunks 5. PBX requirements Order additional hardware or software for your PBX to accommodate new E1 lines. The following telephony components or services can take from 4 to 6 weeks to order and install: 6. Cisco Unified MeetingPlace phone number(s) 7. Digital line requirements Standard E1 Network Telco Service (trunk lines, main phone number, or combined access numbers) PBX specific hardware or software (Smart Blades, software upgrade) Additional hardware (CSU, UPS) The main phone number (800 and/or local) at the beginning of the range. Channelized into 30 channels (30 B channels, plus 1 D-channel and 1 framing channel Ability to receive calls Signaling One of the following: Ability to place calls (required for outdials and alarm outcalls) Ability to hunt all ports (linear, circular, UCD, or ACD) Toll restriction on ports lifted (required for outdials and alarm outcalls) DTMF generated by all PBX or network phones, including the operator console Male RJ-48C jack connector (on the Cisco Unified MeetingPlace side) If using DID Meeting Access, numbering plan for DID/DNIS or DDI, including ranges for Combined, Profile, and Direct Meeting Access Euro-ISDN (default configuration) QSIG: either QSIG_ECMA (channels are numbered 1 30) or QSIG_ETSI (channels are numbered 1 15, 17 31) 3-10
Chapter 3 Worksheet 3-4: Cisco Unified MeetingPlace Telephony Requirements (Europe) Coding format One of the following: HDB3 (strongly recommended) AMI Note If AMI is used, the number of zeros (0) in a row may exceed the recommended Telco value of 15. This can happen if all participants in the same meeting are on the same span. In that case, a 0s pattern can be sent to all channels at the same time, exceeding the maximum. This, in turn, can lead to meeting participants being dropped by the Telco from the meeting. Therefore, use B8ZS or Jammed Bit, if possible. Framing One of the following: CRC4 checking (strongly recommended) Non-CRC4 Additional hardware No channel service unit (CSU) required in Europe, because the Telco provides its own. 8. Modem requirements Analog phone line, or PBX connection Pilot number, accessible from outside RJ-11C connector 9. Translation table Translation table requirements for dialing restrictions. 3-11
Worksheet 3-5: Cisco Unified MeetingPlace LAN Requirements (Worldwide) Chapter 3 Worksheet 3-5: Cisco Unified MeetingPlace LAN Requirements (Worldwide) This worksheet describes the LAN requirements for the United States, Canada, and Japan. 1. Cable/connector requirements 2. Desktop requirements (MeetingTime) Connections from Cisco Unified MeetingPlace server to your network: For twisted-pair Ethernet, 100BASE-TX. Provide an RJ-45 connector. 10BASE-T works, but is not recommended. This cable is customer supplied. Connection from Cisco Unified MeetingPlace Multi Access Blade to your network: For twisted-pair Ethernet, 100BASE-TX. Provide an RJ-45 connector. This cable is supplied by Cisco. Cisco provides a 7.5-meter CAT-5e cable with Ferrite snap-on bead (#3300-0029-02) on one end. If the cable is changed, the snap-on bead must be moved. Note The Ethernet switch port (or any other network devices) that the Multi Access Blade connects to directly must be set to fixed 100BASE-TX Full Duplex. Otherwise, you may experience decreased voice quality. (Set the Ethernet port that connects to the CPU card to auto speed and auto duplex.) Provide a desktop system connected to your LAN with the following minimum configuration: Any standard desktop system that runs on Microsoft Windows 98, NT 4.0 or later, or 2000 15 MB available disk space 16 MB RAM (24 MB RAM for Windows NT or 256 MB RAM for Windows 2000) Network interface card 3. Hostname Name for Cisco Unified MeetingPlace on your network. 4. IP address Address of the Cisco Unified MeetingPlace host on the network. Note When you request your IP address and hostname, ensure that your LAN Manager adds this to the name server(s). 5. Subnet mask Mask that completes the address for the Cisco Unified MeetingPlace host. 6. Broadcast address Address used to broadcast packets on the local LAN segment. 7. Default gateway Address of the integration that will accept and route information to the other networks. 8. SNMP IP address IP address for which traps will be sent for trap communities. 9. Name server Ensure all workstations running MeetingTime use a name server (DNS/WINS or local hosts table). 10. NTP server IP address We recommend that you time synchronize your Cisco Unified MeetingPlace system with a Network Time Protocol (NTP) server. (For more information, see Worksheet 5-8: System Parameters, page 5-38.) 3-12
Chapter 3 Worksheet 3-5: Cisco Unified MeetingPlace LAN Requirements (Worldwide) 11. Broadcast traffic If the rate of broadcast or multicast packet generation on the LAN segment exceeds an average of 40/second, the Ethernet link from Cisco Unified MeetingPlace into the local LAN must be configured for 100 Mbps to avoid congestion of the link. 12. MeetingTime network requirements 13. MeetingTime LAN speed recommendations If broadcast and multicast traffic exceeds 100 packets/second, the Cisco Unified MeetingPlace system should be isolated from the segment using a router. MeetingTime must be able to open a TCP connection on ports 5001 and 5005 to connect to the network and/or conference server. For MeetingTime to access recordings and attachments, the IP address of the conference server must not be translated using a network address translation scheme. To schedule and monitor small (2 10 participants) meetings: Required: 28 kbs Recommended: 40 kbs To monitor medium meetings (11 60 participants): Required: 35 kbs Recommended: 50 kbs To monitor large meetings (61 120 participants): Required: 50 kbs Recommended: 128 kbs 3-13
Worksheet 3-6: Cisco Unified MeetingPlace VoIP Phone Connections Requirements (Worldwide) Chapter 3 Worksheet 3-6: Cisco Unified MeetingPlace VoIP Phone Connections Requirements (Worldwide) This worksheet describes the telephony requirements for Voice over IP (VoIP) telephony connections worldwide. 1. Access ports Number of access ports (user licenses) your company purchased. 2. VoIP access ports Number of these access ports that will be VoIP. 3. Cisco Unified MeetingPlace phone numbers for VoIP access 4. IP addresses to be used by the VoIP RTP streams 5. Subnet mask for RTP IP addresses Note The number of VoIP ports cannot exceed the capacity allowed by your hardware. The main phone numbers (800 and local) at the beginning of the range. IP addresses that will be used by IP phones to connect to the Cisco Unified MeetingPlace Multi Access (MA) blades. Note Up to four IP addresses are needed for Multi Access blades used for VoIP. Each MA-16 needs 1 IP address if 240 or fewer ports are configured on it. For more than 240 ports, an MA-16 needs 2 IP addresses. If the second IP address on an MA-16 is unused, 0.0.0.0 can be used. Each MA-4 will always need exactly 1 IP address. The standard masks used to subdivide the network into smaller groups of IP addresses. Each MA blade requires a subnet mask. For most customers, this mask is the same for all MA blades. 6. Default gateway The IP address of a gateway machine (Cisco MCS) on the local network. Packets with non-local addresses are sent here if no other route is known. Each MA blade requires a default gateway. For most customers, this address is the same for all MA blades. 7. Hostname and IP address of the Cisco Unified MeetingPlace 8100 series server 8. Hostname and IP address of the Cisco Unified MeetingPlace H.323/SIP IP Gateway server 9. (Optional) Hostname and IP address of the Cisco Unified MeetingPlace Web Conferencing server You need to know this value to install and configure Cisco Unified MeetingPlace Audio Server. Configuration of a Cisco Unified MeetingPlace 8100 series server for VoIP involves three components: Cisco Unified MeetingPlace Audio Server Cisco Unified MeetingPlace H.323/SIP IP Gateway software Cisco Unified CallManager or other VoIP soft switch Needed only if the Cisco Unified MeetingPlace Web Conferencing server is running on the same server hardware as Cisco Unified MeetingPlace H.323/SIP IP Gateway software. 3-14
Chapter 3 Worksheet 3-6: Cisco Unified MeetingPlace VoIP Phone Connections Requirements (Worldwide) 10. Hostname and IP address of the VoIP soft switch 11. Special soft switch requirements 12. Network infrastructure connected to Cisco Unified MeetingPlace MA blades set to 100BASE-T Full Duplex Cisco SIP Proxy server, Avaya Communication Manager, or other soft switch. Any soft switch requirements that will affect the Cisco Unified MeetingPlace VoIP installation. Ethernet connections that carry the RTP streams to and from the MA blades must be 100 Mbps with no negotiation on either end. 13. Codec type Will you use G.711 u-law, G.711 A-law, or G.729? 14. Packets per second If using a G.711 codec, the number of packets per second that will be transferred (10, 20, 30). Default is 20 per second. 15. Silence suppression Will silence packets be suppressed? Yes means the number of VoIP packets leaving Cisco Unified MeetingPlace will be typically 2 or 3 per 5 seconds if silence is detected. No means packet usage is typically 50 to 150 packets per 5 seconds (depending on the codec used), which uses more bandwidth. 16. QoS system to be used The Quality of Service (QoS) system to be used. This must be either the IP Precedence system or the Differentiated Services Code Point (DSCP) system. 17. QoS subfields If using IP Precedence, select an IP Precedence value from the following list: 0 routine 1 priority 2 immediate 3 flash 4 flash override 5 CRITIC/ECP (standard value) 6 internetwork control 7 network control Also, select a Type of Service (ToS) value (0 to 15). We recommend 0 If using DSCP, select the DSCP value (0 to 63). Standard value is 40. 18. Base UDP port Each MA VoIP entity (anything that requires an IP address) requires a Base UDP port. The Cisco Unified MeetingPlace Audio Server system provides a default value during configuration. The default is 5000 for the first RTP entity and increases by 1000 for each entity. Although we recommend that you accept these defaults, you can provide different values for your needs. 3-15
Worksheet 3-6: Cisco Unified MeetingPlace VoIP Phone Connections Requirements (Worldwide) Chapter 3 19. Jitter buffer setting Each MA blade requires jitter buffer settings to handle variances in the rate at which VoIP packets are received. The Cisco Unified MeetingPlace Audio Server system provides a default value during configuration. The default initial jitter delay (range 1 1000 msec) is 100 milliseconds. This is a good compromise between audio conversation delay and loss of data. The default jitter optimization value (range 0 12) of 7 determines how quickly the system changes the jitter buffer delay, based on network changes. Although we recommend that you accept these defaults, you can provide different values for your needs. 20. Translation for incoming numbers 21. Translation for outgoing numbers How you want incoming numbers to be translated using the dial groups feature in the Cisco Unified MeetingPlace H.323/SIP IP Gateway server (if at all). How you want phone numbers to be translated using the Cisco Unified MeetingPlace Audio Server translation table feature for calls out of the Cisco Unified MeetingPlace system. 22. Digit transport Determine the method by which digits will be sent to the Cisco Unified MeetingPlace Audio Server system through VoIP by your network: As part of the voice stream (fully in-band) As part of the RTP stream carrying the voice packets but as separate packets (referred to as RFC2833 digits) Sent direct to the MP VoIP Gateway which will relay them to Cisco Unified MeetingPlace (full out-of-band) 23. RFC2833 digits If using RFC2833 packets for digit transport, the Cisco Unified MeetingPlace Audio Server system needs to know if a packet actually holds an RFC2833 digit. There is no standard packet identifier number to indicate an RFC2833 digit. The range for this payload type is 96 to 127. Cisco uses an internal default of 101; however, contact your network administrator for what your network will use. 3-16