Hi, My VSP uses Asterisk to which I'm connected with an ATA. When I receive an inbound call the invite includes the following... v=0 o=root 32218 32218 IN IP4 202.52.129.50 s=session c=IN IP4 202.52.129.50 t=0 0 m=audio 16864 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off ? - ? - a=ptime:20 a=sendrecv My ATA's 200 OK reply after call setup has the following... v=0 o=CMI-SIPUA 13369 0 IN IP4 211.30.XXX.XXX s=SIP CALL c=IN IP4 211.30.XXX.XXX t=0 0 m=audio 20216 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=rtcp:20217 a=silenceSupp:off ? - ? - a=sendrecv Notice there is no "rtpmap:18 G729/8000" in the reply. The call continues fine. Is it right that there is no codec info in the reply and the call continues?
On 12/18/11 12:55 AM, William Scott wrote:> Notice there is no "rtpmap:18 G729/8000" in the reply. > > The call continues fine. > > Is it right that there is no codec info in the reply and the call continues?The value for 18 is defined in some RFC as being G729/8000 so there's no real need to redefine it in the SDP, especially not since the answering party already knows that the initiating party also uses the same value. -- Andreas Sikkema