<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">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. <div>
<br></div><div>Is Anyone able to query codec capability for any endpoints? I would like to know how you do so.</div><div><br></div><div>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</div>
<div><br></div><div><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; color: red; ">OPTIONS sip:991</span><span style="font-size: 10pt; font-family: Arial; ">@<a href="http://192.168.1.4:5060/" target="_blank" style="color: rgb(0, 0, 204); ">192.168.1.4:5060</a> SIP/2.0</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Max-Forwards: 70</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.1.2" target="_blank" style="color: rgb(0, 0, 204); ">sip:asterisk@192.168.1.2</a>>;tag=as1fd2a50c</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">To: <<a href="http://sip:991@192.168.1.4:5060/" target="_blank" style="color: rgb(0, 0, 204); ">sip:991@192.168.1.4:5060</a>></span></p><p class="MsoNormal">
<span style="font-size: 10pt; font-family: Arial; ">Contact: <<a href="http://sip:asterisk@192.168.1.2:5060/" target="_blank" style="color: rgb(0, 0, 204); ">sip:asterisk@192.168.1.2:5060</a>></span></p><p class="MsoNormal">
<span style="font-size: 10pt; font-family: Arial; ">Call-ID: <a href="http://010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060/" target="_blank" style="color: rgb(0, 0, 204); ">010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060</a></span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">CSeq: 102 OPTIONS</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">User-Agent: Asterisk PBX 1.8.6.0</span></p><p class="MsoNormal">
<span style="font-size: 10pt; font-family: Arial; ">Date: Mon, 24 Oct 2011 19:14:47 GMT</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Supported: replaces, timer</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Content-Length: 0</span></p><p class="MsoNormal">
<span style="font-size: 10pt; font-family: Arial; "><br></span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; "></span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; "><--- SIP read from UDP:<a href="http://192.168.1.4:5060/" target="_blank" style="color: rgb(0, 0, 204); ">192.168.1.4:5060</a> ---></span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">SIP/2.0 200 OK</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.1.2" target="_blank" style="color: rgb(0, 0, 204); ">sip:asterisk@192.168.1.2</a>>;tag=as1fd2a50c</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">To: <<a href="http://sip:991@192.168.1.4:5060/" target="_blank" style="color: rgb(0, 0, 204); ">sip:991@192.168.1.4:5060</a>>;tag=003d3418e2fce011b081701a0413e3f3</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Call-ID: <a href="http://010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060/" target="_blank" style="color: rgb(0, 0, 204); ">010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060</a></span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">CSeq: 102 OPTIONS</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Contact: <<a href="http://sip:991@192.168.1.4:5060/" target="_blank" style="color: rgb(0, 0, 204); ">sip:991@192.168.1.4:5060</a>></span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE</span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Server: SIPPER for PhonerLite</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial; ">Content-Length: 0</span></p></div></span>