I have been sending OPTIONS requests both programatically (my own code), manually via SIP VERIFY PEER x and automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice the Asterisk generated OPTIONS does not specify any Accept header (ie Accept=application/sdp). I was thinking maybe that is why I don't get any SDP coming back. My own code generated OPTIONS includes the Accept header and still I see no SDP.
Is Anyone able to query codec capability for any endpoints? I would like to know how you do so. Below is excerpt from the automatic OPTIONS query I see in the sip logs from setting verify=true. No Accept header. Does anyone believe that to be the problem? Notice the response has content length=0 and no SDP. Any ideas appreciated OPTIONS sip:991@192.168.1.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as1fd2a50c To: <sip:991@192.168.1.4:5060> Contact: <sip:asterisk@192.168.1.2:5060> Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Mon, 24 Oct 2011 19:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <--- SIP read from UDP:192.168.1.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169 From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as1fd2a50c To: <sip:991@192.168.1.4:5060>;tag=003d3418e2fce011b081701a0413e3f3 Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060 CSeq: 102 OPTIONS Contact: <sip:991@192.168.1.4:5060> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0
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