[cisco-voip] Codec negotiation issue, a little strange

[Brian Meade]

CVP doesn't change SDP info and CUSP might not either, don't remember.  Can
you run "debug ccsip messages" on the H.323 GW to see if it's offering
anything other than G.711ulaw in the Invite to CUSP?  If it's only G.711
there, it must be getting changed by CUSP.  I'm not familiar enough with it
to know what to check though.

On Wed, May 6, 2015 at 5:40 PM, Ryan Huff <ryanhuff at outlook.com> wrote:

> I have a situation, where, in some case the ingress pstn call leg is
> trying to use g.729 (when there is nothing in the gateway that would
> indicate it's preference).
>
> Call path when G729 is negotiated:
>
> PSTN -> h.323(PRI) dial-peer match -> SIP trunk to Unified Proxy Server -
> > SIP Customer Voice Portal -> SIP To vXML Server -> Send SIP invite to
> call manager and the SDP contains G729
>
> Call path when G711 is negotiated:
>
> PSTN - >h .323(PRI) dial-peer match -> CCM -> ring phone
>
> - The gateway and IP phone are in the same region and the region is
> related to itself with G711.
> - The region of the DP of the SIP trunk is related to the region of the
> gateway and the region of the phone with G711
> - The voice class codec on the router only has g711 set as the 1st
> preference
> - The matched dial-peer (h.323) for the SIP trunk to CUSP is specifying
> the voice class codec correctly
> - The IP phone is a 7962/42
>
> My question is how and why is the far end negotiating G729? This site does
> have limited CIR (10 Mbps) and the CVP/vXML servers are in a different
> geographic location than the gateway/IP phone.
>
> My thought is that since the IP phone can negotiate G.729, it could be a
> bandwidth thing where it is just choosing to use the lower bandwidth codec
> BUT the invite to CCM is coming from CVP. The SDP in the invite shows G729.
>
> Content-Type: application/sdp
> App-Info: <*vXMLVoiceServerIPAddress*:8000:8443>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 7410 5486 IN IP4 *GatewayIpAddress*
> s=SIP Call
> c=IN IP4 *GatewayIpAddress*
> t=0 0
> m=audio 19808 RTP/AVP 0 18 100 101
> c=IN IP4 *GatewayIpAddress*
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:100 X-NSE/8000
> a=fmtp:100 192-194
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20150506/abdb7d2c/attachment.html>