I friggin' hate dial-peer 0; that is for dang sure. When doing "sh dialplan num XXXXXXXXXX" it pops the first peer I expect it to.
That is a good thought; worth a little research on. Maybe I use a peer with an exact match pattern; maybe that will do something different. Thanks for that, new angle and I like it. Thanks, Ryan From: [email protected] To: [email protected]; [email protected] CC: [email protected] Date: Wed, 6 May 2015 15:24:19 -0700 Subject: RE: [cisco-voip] Codec negotiation issue, a little strange I have vague recollections of similar issues when an incoming call wasn’t properly matching a dial peer – it was hitting a “default” which I believe was g.729. I’d verify it’s actually hitting the dial-peer you think it should be. From: cisco-voip [mailto:[email protected]] On Behalf Of Ryan Huff Sent: Wednesday, May 6, 2015 5:15 PM To: Brian Meade Cc: [email protected] Subject: Re: [cisco-voip] Codec negotiation issue, a little strange I'll do some testing after hours. When the client reported the issue, I just threw a transcoder in the MRGL for the SIP trunk in call manager to 'plug the leak'. I'll let you know what I find and/or if I can find out if CUSP is screwing with me. Thanks, RyanDate: Wed, 6 May 2015 17:54:57 -0400 Subject: Re: [cisco-voip] Codec negotiation issue, a little strange From: [email protected] To: [email protected] CC: [email protected] 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 <[email protected]> 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 [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip
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