Cisco Cisco IP Contact Center Release 4.6.2 Leaflet

To mitigate the bandwidth demands, use any combination of the following options: 

Use fewer call and ECC variables on the child PG.

Certain messages transmit call data from the child Unified CCE system to the parent. Reducing the 
size and quantity of variables used will reduce the data transmitted for these events. (See the Note 
under 

.)

Use the MAPVAR = IIIIIIIIII and MAPECC = IIIIIIIIII peripheral configuration parameters.

If you do not use the MAPVAR and MAPECC option (which means that the settings default to 
MAPVAR = BBBBBBBBBB and MAPECC = BBBBBBBBBB), then for every ROUTE_SELECT 
sent to the child, all Call and ECC variables used on the parent are also sent to the child. If you use 
the I (Import) or N (None) option for MAPVAR, MAPECC, or both, then the Gateway PG will not 
send these variables over the line to the child system. If a lot of call variables and/or ECC variables 
are used on the parent, these parameter settings can save some bandwidth.

Eliminating Import (I or B setting) of data does not save any bandwidth because, even though the 
Gateway PG does not import the data, the child Unified CCE system still transmits it.

There are many factors to consider when assessing the traffic and bandwidth requirements for Agent and 
Supervisor Desktops in a Unified CCE environment. While the VoIP packet stream bandwidth is the 
predominant contributing factor to bandwidth, other factors such as call control, agent state signaling, 
silent monitoring, recording, and statistics must also be considered.

VoIP packet stream bandwidth requirements are derived directly from the voice codec deployed (G.729, 
G.711, and so forth), and can range from 4 kbps to 64 kbps per voice stream. Therefore, the contact 
center's call profile must be well understood because it defines the number of straight calls (incoming or 
outgoing), consultative transfers, and conference calls, and consequently the number of VoIP packet 
streams, that are active on the network. In general, the number of VoIP packet streams will be typically 
slightly greater than one per agent, to account for held calls, silent monitoring sessions, active 
recordings, consultative transfers, and conference calls.

Call control, agent state signaling, silent monitoring, recording, and statistics bandwidth requirements 
can collectively represent as much as 25% to 50% of total bandwidth utilization. While VoIP packet 
stream bandwidth calculations are fairly straightforward, these other factors depend heavily upon 
implementation and deployment details and are therefore discussed further in the sections below.

Because WAN links are usually the lowest-speed circuits in an Cisco Unified Communications network, 
attention must be given not only to bandwidth, but also to reducing packet loss, delay, and jitter where 
voice traffic is sent across these links. G.729 is the preferred codec for use over the WAN because the 
G.729 method for sampling audio introduces the least latency (only 30 ms) in addition to any other 
delays caused by the network. The G.729 codec also provides good voice quality with good compression 
characteristics, resulting in a relatively low (8 kbps) bandwidth utilization per stream.