Hello, We are using a Cisco 827-4V as a SIP gateway for an analog phone. The phone registers, but we're getting one way audio. We found there is an async routing path. The PSTN GW should be talking to the SIP NAT server, but the Cisco is sending two "contact information"s in the SDP. The NAT server updates the first CI with its IP, but the Application Server ignores that one and sends the second entry with the phone IP to the PSTN GW.
My question is - isn't this a violation of the RFC having two contact information vaules in an SDP message? I have querried the net, and found a couple examples with this, but none on Cisco's web page. Does anyone know how to eliminate one of the entries? The Cisco config is very basic, just a codec definition, and the SIP register server. Here is a copy of the SDP from a packet capture.
v=0 o=CiscoSystemsSIP-GW-UserAgent 6043 2386 IN IP4 192.168.1.19 s=SIP Call c=IN IP4 192.168.1.19 t=0 0 m=audio 17266 RTP/AVP 18 101 c=IN IP4 192.168.1.19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active