Cisco Cisco Customer Voice Portal 8.0(1) Release Notes

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Release Notes for Cisco Unified Customer Voice Portal, Release 7.0(2) 
 
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Note: On earlier versions of CUPS where the "record route header" feature can be and is 
disabled, this issue won't occur even when using UDP. 
Note: Another workaround is to point CUCM SIP Trunk direct to Unified CVP instead of 
CUPS, and UDP may be retained.  This is the trunk used with MTP regarding 
CSCsm12716 mentioned above. 
Note: TCP may automatically be used on a CUPS call if the MTU size of the SIP message 
exceeds 1300 bytes by default.  This generally happens when signaling forward 
unconditional is used on the ingress gateway.  To preserve UDP transport usage you may 
change CUP service parameter setting for MTU upconversion to 1800 instead of 1300. 
In a certain scenario, basic video calls from a Telepresence endpoint will require MTP to 
be enabled on the SIP trunk pointing to Unified CVP. The scenario is as follows: 
1.  Telepresence caller dials into Unified CVP 
2.  receives audio-only IVR 
3.  caller is then is transferred to an Agent on an audio-only IP Phone 
MTP must be enabled on the SIP trunk or else one-way audio is encountered.  (Caller 
could not hear audio without MTP enabled)   
For Unified CVP 4.1 and later releases, an MTP is no longer needed for calls originated 
by Unified CM H.323 trunks. However, the Unified CVP Configuration and 
Administration Guide still includes instructions for configuring MTP for these scenarios.  
In Chapter 2, the section "Calls Which are Originated by Unified CM" states the 
requirement that MTP be used. You do not need to configure MTP in this scenario. 
In Chapter 10, the section "ICME warm Consult/Conference" also provides instructions 
for configuring MTP.  You do not need to configure MTP in this scenario. 
Note there are still some instances where you need to configure MTP, such as for Video 
calls and for certain workarounds. For example, see "MTP and completing transfers while 
still in queue" later in these release notes. 
If the agent completes the transfer to caller while still in queue, it causes the mid call 
media change for the voice browser on the VXML gateway, which the gateway doess not 
currently support (CSCsm12716). Therefore, MTP allocation becomes necessary.