J.R. Pauley
2011-Oct-26 10:11 UTC
[asterisk-users] OPTIONS to determine codec capability before an INVITE
I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)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 in both 2 and 3 examples 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. The rfc says the ACCEPT SHOULD be present so I'm thinking that is a Asterisk bug perhaps. In example 1 My own UAC code generated OPTIONS includes the Accept header yet still I see no SDP coming back from endpoints. I have tried using X-lite and PhonerLite softclients. I'm hoping there is a simple explanation or something I can do. Is Anyone able to query codec capability for any endpoints outside of a normal INVITE? 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 at 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 at 192.168.1.2>;tag=as1fd2a50c To: <sip:991 at 192.168.1.4:5060> Contact: <sip:asterisk at 192.168.1.2:5060> Call-ID: 010fdb653903a2022b99ed1d40c0b8db at 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 at 192.168.1.2>;tag=as1fd2a50c To: <sip:991 at 192.168.1.4:5060>;tag=003d3418e2fce011b081701a0413e3f3 Call-ID: 010fdb653903a2022b99ed1d40c0b8db at 192.168.1.2:5060 CSeq: 102 OPTIONS Contact: <sip:991 at 192.168.1.4:5060> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111026/e85cb53d/attachment.htm>