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New Member

SIP Call flow, for SIP trunk

We have integrated our CUCM 8.0 with Third-Party callcenter solution over SIP trunk. We are facing intermittent one way audio for the calls made from Third-party client, which is installed on the agent PC, to the PSTN.

Third-partyClient on PC > SIP Trunk > CUCM > Voice Gateway > PSTN

From the packet capture we can see that for the non-working calls the RTP coming from the Voice Gateway and for working call RTP coming from the CUCM (Snapshot from packet capture attached).

But we know for the SIP call flow, once the initial signalling is done between the servers (Here CUCM and Third-party server) the RTP streams will be send only between the end-points. Can any one brief me what's happening here. Appreciate your response.

  • CUCM Pub -
  • CUCM Sub1 -
  • Voice Gateway -
  • Third-party server -

Packet capture can be downloaded from the dropbox link :

Cisco Employee

Hi,The reason for this could


The reason for this could be that a software MTP is getting allocated for a working call, so the RTP is passing through the server to which the MTP is registered. You can check the MTP usage through RTMT for working calls to confirm this.



New Member

I would agree with Manish.

I would agree with Manish. Try using "Use Trusted Relay Point" on the SIP Trunk, and tick "Trusted Relay Point" on your software MTP.



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