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

[Ryan Huff]

Brian,

I'm guessing that the router in this case is somehow, messing with the codec.

Since the regions are g722/g711 that means g729 would be allowed in the region because it fits the bandwidth profile of the region.

Do you think doing a "pass-thru content unsupp" would be helpful in either eliminating or proving the router as the source of the issue?

As I understand it, this should allow sdp to pass to the other call leg without media negotiations.

Thanks,

Ryan

-------- Original Message --------
From: Brian Meade <bmeade90 at vt.edu>
Sent: Wednesday, May 6, 2015 05:54 PM
To: Ryan Huff <ryanhuff at outlook.com>
Subject: Re: [cisco-voip] Codec negotiation issue, a little strange
CC: cisco-voip at puck.nether.net

>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
>>
>>
>>
>> _______________________________________________
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>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
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