Cisco Cisco Customer Voice Portal 8.0(1) Design Guide
9-9
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
OL-15989-06
Chapter 9 Network Infrastructure Considerations
Call Admission Control
Voice Traffic (G.711 and G.729)
Unified CVP can support both G.711 and G.729. However, both call legs and all IVR on a given call
must use the same voice codec. If you are using ASR/TTS for speech recognition, then G.711 must be
used because ASR/TTS servers support only G.711. For the most current bandwidth information on
voice RTP streams, refer to the latest version of the Cisco Unified Communications SRND Based on
Cisco Unified Communications Manager, available at
must use the same voice codec. If you are using ASR/TTS for speech recognition, then G.711 must be
used because ASR/TTS servers support only G.711. For the most current bandwidth information on
voice RTP streams, refer to the latest version of the Cisco Unified Communications SRND Based on
Cisco Unified Communications Manager, available at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides
_list.html
_list.html
Call Admission Control
Call admission control is the mechanism for determining if there is enough bandwidth available on the
network to carry an RTP stream. Unified CM can use its own locations mechanism or RSVP to track
bandwidth between the ingress gateway and destination IP phone locations.
network to carry an RTP stream. Unified CM can use its own locations mechanism or RSVP to track
bandwidth between the ingress gateway and destination IP phone locations.
For more information about call admission control, see the chapter on
.
Note
RSVP. Cisco Unified CM 5.0 introduced support for Resource Reservation Protocol (RSVP) between
endpoints within a cluster and Unified CM Release 8.0 introduces RSVP over the SIP trunk. RSVP is a
protocol used for call admission control, and it is used by the routers in the network to reserve bandwidth
for calls. RSVP is not qualified for call control signaling via the Unified CVP Call Server in SIP or H.323
in the 8.0(1) release. The recommended solution for Call Admission Control is to employ Locations
configuration on Unified CVP and in Unified CM.
endpoints within a cluster and Unified CM Release 8.0 introduces RSVP over the SIP trunk. RSVP is a
protocol used for call admission control, and it is used by the routers in the network to reserve bandwidth
for calls. RSVP is not qualified for call control signaling via the Unified CVP Call Server in SIP or H.323
in the 8.0(1) release. The recommended solution for Call Admission Control is to employ Locations
configuration on Unified CVP and in Unified CM.
For more information on RSVP, refer to the latest version of the Cisco Unified Communications SRND
Based on Cisco Unified Communications Manager, available at
Based on Cisco Unified Communications Manager, available at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides
_list.html
_list.html
Local Branch Call Admission Control (LBCAC/Queue-at-the-Edge)
When you are using the Unified CVP Branch Office call flow model deployment you need to control the
number of calls that go over the WAN link to branch offices, based on the available bandwidth of the
WAN link. Decisions for admitting calls are based on the Call Admission Control (CAC) computations
which must be correct and representative of the bandwidth being used by an individual call. These
computations must work whether the calls are IP call between two phones within CCM, calls over
SIP/H.323 trunks, or calls originated from TDM-IP GW.
number of calls that go over the WAN link to branch offices, based on the available bandwidth of the
WAN link. Decisions for admitting calls are based on the Call Admission Control (CAC) computations
which must be correct and representative of the bandwidth being used by an individual call. These
computations must work whether the calls are IP call between two phones within CCM, calls over
SIP/H.323 trunks, or calls originated from TDM-IP GW.
Additionally, for queue-at-the-edge functionality, the call originating from a specific branch office
should be deterministically routed to a local VXML Gateway based on priority. That is, always choose
a local branch agent if possible.
should be deterministically routed to a local VXML Gateway based on priority. That is, always choose
a local branch agent if possible.
The following diagram illustrates a typical branch office deployment.