Cisco Unified MeetingPlace, Release 6.x -- Worksheet 3-6: VoIP Phone Connections Requirements (Worldwide)

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Cisco Unified MeetingPlace, Release 6.x > Cisco Unified MeetingPlace Audio Server > Planning the installation > Telephony and LAN Planning > Telephony and LAN Planning Worksheets




This worksheet describes the telephony requirements for Voice over IP (VoIP) telephony connections worldwide.


Task Description Complete

Access ports

Number of access ports (user licenses) your company purchased.

 

VoIP access ports

Number of these access ports that will be VoIP.

Note: The number of VoIP ports cannot exceed the capacity allowed by your hardware.

 

Cisco Unified MeetingPlace phone numbers for VoIP access

The main phone numbers (800 and local) at the beginning of the range.

 

IP addresses to be used by the VoIP RTP streams

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.

 

Subnet mask for RTP IP addresses

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.

 

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.

 

Hostname and IP address of the Cisco Unified MeetingPlace 8100 series server

You need to know this value to install and configure Cisco Unified MeetingPlace Audio Server.

 

Hostname and IP address of the Cisco Unified MeetingPlace H.323/SIP IP Gateway 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
 

(Optional) Hostname and IP address of the Cisco Unified MeetingPlace Web Conferencing server

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.

 

Hostname and IP address of the VoIP soft switch

Cisco SIP Proxy server, Avaya Communication Manager, or other soft switch.

 

Special soft switch requirements

Any soft switch requirements that will affect the Cisco Unified MeetingPlace VoIP installation.

 

Network infrastructure connected to Cisco Unified MeetingPlace MA blades set to 100BASE-T Full Duplex

Ethernet connections that carry the RTP streams to and from the MA blades must be 100 Mbps with no negotiation on either end.

 

Codec type

Will you use G.711 u-law, G.711 A-law, or G.729?

 

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.

 

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.

 

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.

 

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.

 

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.

 

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.

 

Translation for incoming 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).

 

Translation for outgoing numbers

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.

 

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)
 

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.

 

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