I'm doing SIP trunking with the german voice provider QSC. You can find the config and the version data below. In some points it works fine, but dialing in from somewhere else (e.g. mobile) than from a fixed line of the Deutsche Telekom doesn't work. I get messages and errors about failed media negotiation. But I can't see any difference in the INVITE message as compared to successful calls. Can you please tell me what goes wrong and how to fix it? Attached are files with debugs (messages, media, error) of both the unsuccessful and the successful communication, and an on-the-wire trace of the unsuccessful communication. Thanks.
This problem is likely due to codec interworking between the external SIP device and the version of IOS on the router. The call is getting rejected due to a codec mismatch. Since you do not have any 'incoming called-number' statement, the router would use the default dial peer which specifies G729.
There is a reasonably well known issue involving codec interworking that is due to Cisco defect CSCeh19375.
This DDTS was not specifically a bug, but it added IOS compliance for RFC 3555. RFC 3555 changes the way G729 is dealt with - the parameter "annexb" must be included in the SDP. In this case, the parameter is not included in the SDP from the SIP proxy and as a result, the router will try and use G729br8. This codec is not configured and the call fails.
As a fix, try adding G729br8 codec to a voice class codec and binding this to dial peer 10 by means of the command 'incoming called-number .'
voice class codec 1
codec preference 1 g729br8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
incoming called-number . <---- add this
voice-class codec 1 <---- add this
translation-profile outgoing outward
session protocol sipv2
session target sip-server
This will match the correct codec when the SIP Proxy attempts to use G729.
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