<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">There is actually a codec list but those are static RTP payload types.<div>You are offering PCMU, PCMA, RFC2833 DTMF, and Comfort Noise </div><div><br></div><div>See:</div><div><a href="http://www.iana.org/assignments/rtp-parameters">http://www.iana.org/assignments/rtp-parameters</a></div><div><br></div><div>If you need the full SDP for even known static types set verbose_sdp=true in vars.xml.</div><div><a href="http://wiki.freeswitch.org/wiki/Variable_verbose_sdp">http://wiki.freeswitch.org/wiki/Variable_verbose_sdp</a></div><div><br></div><div>Spencer</div><div><br></div><div><br></div><div><div><div>On May 24, 2011, at 7:56 PM, Liam Farr wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Hi,
<div><br></div><div>My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?</div><div><br></div><div><br></div><div><div>send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:</div>
<div> ------------------------------------------------------------------------</div><div> INVITE <a href="http://sip:0800000000@1.2.3.4:5060/">sip:0800000000@1.2.3.4:5060</a> SIP/2.0</div><div> Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr</div>
<div> Max-Forwards: 69</div><div> From: "Liam - Test" <<a href="mailto:sip%3A12345678@9.8.7.6">sip:12345678@9.8.7.6</a>>;tag=ryFQea59e6Upe</div><div> To: <<a href="http://sip:0800000000@1.2.3.4:5060/">sip:0800000000@1.2.3.4:5060</a>></div>
<div> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43</div><div> CSeq: 12805602 INVITE</div><div> Contact: <<a href="http://sip:12345678@9.8.7.6:5060/">sip:12345678@9.8.7.6:5060</a>></div><div> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported</div>
<div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY</div><div> Supported: timer, precondition, path, replaces</div><div> Allow-Events: talk, hold, refer</div><div> Content-Type: application/sdp</div>
<div> Content-Disposition: session</div><div> Content-Length: 189</div><div> X-FS-Support: update_display</div><div> Remote-Party-ID: "Liam - Test" <<a href="mailto:sip%3A12345678@9.8.7.6">sip:12345678@9.8.7.6</a>>;party=calling;screen=yes;privacy=off</div>
<div><br></div><div> v=0</div><div> o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6</div><div> s=FreeSWITCH</div><div> c=IN IP4 9.8.7.6</div><div> t=0 0</div><div> m=audio 18646 RTP/AVP 0 8 101 13</div><div>
a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> ------------------------------------------------------------------------</div></div><div><br></div><div><br></div><div>Thanks</div><div><br></div>
<div>Liam</div>
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