[asterisk-users] OPTIONS to determine codec capability before an INVITE

[J.R. Pauley]

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
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