Cisco Cisco Customer Voice Portal 8.0(1) Design Guide

Page of 223
 
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.
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. 
More specifically, the Call-Info header in the SIP INVITE will specify the originating device in the 
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.
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.
For more information on RSVP, refer to the latest version of the Cisco Unified Communications SRND 
Based on Cisco Unified Communications Manager
, available at 
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.
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 
.) 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 
 sends a call to the.3 server, the call is 
rejected.