Cisco Cisco Unified Contact Center Management Portal 8.5 Release Note
Chapter 4 Unified Contact Center Enterprise Desktop
Cisco Unified Contact Center Enterprise 8.x SRND
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For bandwidth usage information, see the
There must be sufficient bandwidth available from the agent IP phone to the RSM server to support the
monitoring voice stream, in addition to the regular voice streams for the call. This is important for agents
who work remotely, at home and small branches on limited bandwidth or WAN connectivity. Regular Call
Admission Control (CAC) and bandwidth calculations are applicable for monitoring calls.
monitoring voice stream, in addition to the regular voice streams for the call. This is important for agents
who work remotely, at home and small branches on limited bandwidth or WAN connectivity. Regular Call
Admission Control (CAC) and bandwidth calculations are applicable for monitoring calls.
Since G.711 is the only codec supported for monitoring calls between agent IP phone and RSM server
(phonesim), use the
(phonesim), use the
planning.
Agent Phone Transcoding Implications in G729 Environments
The monitoring call established between RSM’s simulated phone (simphone) and Agent’s phone is subject
to regular call admission control (CAC) procedures. RSM server (phonesim) supports only G.711 codec
and hence the simphones should be configured in a region (UCM Region) for G.711 only. Due to this, the
monitoring call always negotiates a G.711 codec between RSM’s simphone and Agent phone’s BiB. If the
Agent-Customer conversation is in G.711 then no transcoding is required. If the Agent-Customer
conversation is in G.729 then transcoding is performed by Agent Phone’s BiB itself. No additional
transcoding resources or hardware needed in Voice Gateway. This BiB transcoding capability exists in both
physical and soft-phone (IPC 7.0 or higher) models.
to regular call admission control (CAC) procedures. RSM server (phonesim) supports only G.711 codec
and hence the simphones should be configured in a region (UCM Region) for G.711 only. Due to this, the
monitoring call always negotiates a G.711 codec between RSM’s simphone and Agent phone’s BiB. If the
Agent-Customer conversation is in G.711 then no transcoding is required. If the Agent-Customer
conversation is in G.729 then transcoding is performed by Agent Phone’s BiB itself. No additional
transcoding resources or hardware needed in Voice Gateway. This BiB transcoding capability exists in both
physical and soft-phone (IPC 7.0 or higher) models.
For more information, see the “Codec for Monitoring and Recording Calls” topic in “Monitoring and
Recording” chapter of the
Recording” chapter of the
Failover Redundancy and Load Balancing
Load balancing support is defined as the act of multiple RSM servers being associated together so that the
incoming request load is distributed among them. The definition of failover is multiple RSM servers being
associated together so that if one fails, the other(s) can act in its place. In the future, RSM will support load
balancing and failover with both the Unified CVP and IP IVR VRUs. Currently, this support is not
available in RSM 1.0. RSM 1.0 does, however, support the deployment of multiple standalone RSM
servers within a single Unified CCE environment, and this concept is demonstrated in the advanced
deployment scenarios described in this document.
incoming request load is distributed among them. The definition of failover is multiple RSM servers being
associated together so that if one fails, the other(s) can act in its place. In the future, RSM will support load
balancing and failover with both the Unified CVP and IP IVR VRUs. Currently, this support is not
available in RSM 1.0. RSM 1.0 does, however, support the deployment of multiple standalone RSM
servers within a single Unified CCE environment, and this concept is demonstrated in the advanced
deployment scenarios described in this document.
Table 5 indicates how a failure of each of the various components affects a live supervisor call.
Table 5
Impact of Failures on a Supervisor Call
Component That Fails Worst Possible Impact
VRU Node (IP IVR,
Unified CVP)
Unified CVP)
Supervisor's call is terminated as any VRU failover occurs (depends).
Supervisor may dial back in and log in again once VRU failover is complete
and/or the original failed VRU is working again.
Supervisor may dial back in and log in again once VRU failover is complete
and/or the original failed VRU is working again.