Cisco Cisco Customer Voice Portal 8.0(1) Design Guide
3-10
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
OL-15989-06
Chapter 3 Distributed Deployments
Call Admission Control Considerations
SIP Call Flows
With SIP-based call flows, Cisco Unified CM 6.0 (and prior releases) is able to look at only the source
IP address of the incoming SIP INVITE from Unified CVP. This limitation causes a problem with call
admission control because Unified CM is not able to identify which gateway behind Unified CVP
originated the call.
IP address of the incoming SIP INVITE from Unified CVP. This limitation causes a problem with call
admission control because Unified CM is not able to identify which gateway behind Unified CVP
originated the call.
Cisco Unified CM 6.1 has enhanced the SIP Trunk to look beyond the source IP address and to inspect
information contained in the SIP header when determining which device originated a call. This
enhancement allows the SIP trunk to be dynamically selected by the original source IP address rather
than the remote port on Unified CVP, and therefore different SIP profiles and settings can be used on the
source trunks that are different from the Unified CVP trunk.
information contained in the SIP header when determining which device originated a call. This
enhancement allows the SIP trunk to be dynamically selected by the original source IP address rather
than the remote port on Unified CVP, and therefore different SIP profiles and settings can be used on the
source trunks that are different from the Unified CVP trunk.
More specifically, the Call-Info header in the SIP INVITE will specify the originating device in the
following format:
following format:
<sip: IPAddress:port>;purpose=x-c isco-origIP
Where IPAddress:port indicates the originating device and its SIP signaling port.
This source IP SIP trunk selection feature does not impact the bandwidth monitoring for call admission
control. In Unified CM release 8.0, bandwidth monitoring is performed with SIP using locations
configuration on Unified CVP and Unified CM. The following header is used by the location server in
Unified CM to manipulate bandwidth information for call admission control.
control. In Unified CM release 8.0, bandwidth monitoring is performed with SIP using locations
configuration on Unified CVP and Unified CM. The following header is used by the location server in
Unified CM to manipulate bandwidth information for call admission control.
Call-Info: [urn:x-cisco-remotecc:callinfo];x-cisco-loc-id="PKID";x-cisco-loc-name="Loc-NAME"
RSVP
Cisco Unified CM 5.0 introduced support for Resource Reservation Protocol (RSVP) between endpoints
within a cluster, and 8.0 UCM 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 H323 in the 8.0 release.
The recommended solution for CAC is to employ Locations configuration on Unified CVP and in
Unified CM.
within a cluster, and 8.0 UCM 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 H323 in the 8.0 release.
The recommended solution for CAC 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
H.323 Gatekeeper Call Routing
For proper configuration of remote H.323 gateways with a Unified CM cluster, first consider the H.225
implications of this configuration without the use of gatekeeper.
implications of this configuration without the use of gatekeeper.
When configuring dial-peer destinations for the Cisco IOS Gateways, you must configure a dial peer
pointing to the IP addresses of the Unified CM servers that are processing calls for that gateway. These
server IP addresses must be the same servers that are in the redundancy group of the device pool
definition for that gateway in the Unified CM configuration. (See
pointing to the IP addresses of the Unified CM servers that are processing calls for that gateway. These
server IP addresses must be the same servers that are in the redundancy group of the device pool
definition for that gateway in the Unified CM configuration. (See
.) If the remote H.323
gateway sends a call to a Unified CM server that is not in the redundancy group for that gateway, the call
is rejected. For example, if the Madison gateway in
is rejected. For example, if the Madison gateway in
sends a call to the.3 server, the call is
rejected.