Cisco Cisco Unified MeetingPlace Express 2.1 White Paper
Cisco Systems
Network Provisioning and Configuration
Page 4
2.1.1 B a n d w i d t h Se t t i n g s
Cisco MeetingPlace Express supports only G.711 (mu-law or a-law). Within the core
network, Cisco recommends allowing the end points to use G.711 when communicating
with Cisco MeetingPlace Express, without employing a transcoder.
Assuming the default packet size of 20ms, each voice endpoint can send and receive
approximately 84Kbs of constant bandwidth traffic, including packet overhead. On a 120
port system, that is an aggregate of approximately 10Mbs each direction at the server.
For acceptable voice quality, it is important that this bandwidth always be available, with
negligible packet loss or delay, even when sharing links with high bandwidth applications
such as web conferencing.
Using Voice Activity Detection (VAD) can reduce the average bandwidth consumed by
the voice links. Cisco MeetingPlace Express is compatible with VAD on incoming
streams. It can also be configured to use VAD on outgoing streams. Note, however, that
outgoing streams from conferences are much less frequently silent than they are in two-
way conversations, so the benefit to enabling VAD in Cisco MeetingPlace Express is not
substantial – and enabling it will slightly reduce voice quality.
network, Cisco recommends allowing the end points to use G.711 when communicating
with Cisco MeetingPlace Express, without employing a transcoder.
Assuming the default packet size of 20ms, each voice endpoint can send and receive
approximately 84Kbs of constant bandwidth traffic, including packet overhead. On a 120
port system, that is an aggregate of approximately 10Mbs each direction at the server.
For acceptable voice quality, it is important that this bandwidth always be available, with
negligible packet loss or delay, even when sharing links with high bandwidth applications
such as web conferencing.
Using Voice Activity Detection (VAD) can reduce the average bandwidth consumed by
the voice links. Cisco MeetingPlace Express is compatible with VAD on incoming
streams. It can also be configured to use VAD on outgoing streams. Note, however, that
outgoing streams from conferences are much less frequently silent than they are in two-
way conversations, so the benefit to enabling VAD in Cisco MeetingPlace Express is not
substantial – and enabling it will slightly reduce voice quality.
2.1.2 Qu a l i t y o f Se r v i c e ( QOS)
Cisco MeetingPlace Express supports DSCP marking of voice packets. Assuming that the
server and the network are properly configured to support DSCP QOS, this has the effect
of prioritizing voice packets higher than most other classes of traffic, so that they can get
though even when network links are otherwise overloaded.
Some QOS considerations:
server and the network are properly configured to support DSCP QOS, this has the effect
of prioritizing voice packets higher than most other classes of traffic, so that they can get
though even when network links are otherwise overloaded.
Some QOS considerations:
•
You can configure the server DSCP setting in the Cisco MeetingPlace Express
Administration Center under the Call Configuration / Voice Parameters section.
The default is Expedited Forwarding (EF), which is the Cisco standard for voice.
The default is Expedited Forwarding (EF), which is the Cisco standard for voice.
•
It is common for switches to remark packets from untrusted endpoints to disable
DSCP settings. Be sure that packets from your phones and Cisco MeetingPlace
Express are not remarked. This may require special configuration of the switch
port connected to Cisco MeetingPlace Express.
Express are not remarked. This may require special configuration of the switch
port connected to Cisco MeetingPlace Express.
•
Cisco MeetingPlace Express only marks voice data packets (Routing Table
Protocol [RTP] and RTP Control Protocol [RTCP]) for DSCP. Signaling packets
(SIP or H.323) are not marked and consequently have ordinary priority. Signaling
over congested links can result in call setup failures or delays even if the voice
connections work fine once established.
(SIP or H.323) are not marked and consequently have ordinary priority. Signaling
over congested links can result in call setup failures or delays even if the voice
connections work fine once established.
2.1.3 La t e n c y
To support high quality conversations, end-to-end latency (“mouth to ear”) should not
exceed 150ms. A conference call involves two network links and a conference bridge, as
opposed to just a single network link for an ordinary phone call, so latency will normally
be higher.
exceed 150ms. A conference call involves two network links and a conference bridge, as
opposed to just a single network link for an ordinary phone call, so latency will normally
be higher.