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--></style></head><body lang="EN-NZ" link="blue" vlink="purple"><div class="WordSection1"><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Hi,</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I set <action application="export" data="verbose_sdp=true" /> in my dial plan and this fixed it.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">My provider is rather strict about going through a testing / certification process for any “new” hardware that they connect to their network, and wanted to see the full codec list in the SIP invite, (even though as you say it’s not required under the RFC’s).</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks</span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Liam</span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span></p>
<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Steven Ayre<br>
<b>Sent:</b> Wednesday, 25 May 2011 6:22 p.m.<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Outbound SIP invites without codec list</span></p><p class="MsoNormal"> </p><div><p class="MsoNormal" style="margin-bottom:12.0pt">
"m=audio 18646 RTP/AVP 0 8 101 13"<br><br>That is the codec list.<br> 0 = PCMU (G711 u-law)<br> 8 = PCMA (G711 a-law)<br> 101 = DTMF RFC2833<br> 13 = Comfort Noise<br><br>Numbers 96-127 are dynamic. Anything else is static and defined by IANA (<a href="http://www.iana.org/assignments/rtp-parameters">http://www.iana.org/assignments/rtp-parameters</a>).<br>
<br>A dynamic payload type needs a rtpmap line to map the number to the codec name. That is not required for static payload types, because they have a preset rtpmap.<br><br>FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The advantage of doing so is it makes your packets smaller - less bandwidth usage, and less likely to get a fragmented packet.<br>
<br>Some (very few) devices are broken though and require the rtpmap lines for all codecs. These have broken SIP implementations because the RFC states the rtpmap is optional for static types. There is a compatibility option if you're using one of those and really do require it, but changes are you don't: <a href="http://wiki.freeswitch.org/wiki/Variable_verbose_sdp">http://wiki.freeswitch.org/wiki/Variable_verbose_sdp</a><br>
<br>-Steve<br><br><br></p></div><div><p class="MsoNormal">On 25 May 2011 01:56, Liam Farr <<a href="mailto:liam@intersys.co.nz">liam@intersys.co.nz</a>> wrote:</p><p class="MsoNormal">Hi, </p><div><p class="MsoNormal">
</p></div><div><p class="MsoNormal">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?</p></div><div><p class="MsoNormal"> </p></div><div>
<p class="MsoNormal"> </p></div><div><div><p class="MsoNormal">send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:</p></div><div><p class="MsoNormal"> ------------------------------------------------------------------------</p>
</div><div><p class="MsoNormal"> INVITE <a href="http://sip:0800000000@1.2.3.4:5060" target="_blank">sip:0800000000@1.2.3.4:5060</a> SIP/2.0</p></div><div><p class="MsoNormal"> Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr</p>
</div><div><p class="MsoNormal"> Max-Forwards: 69</p></div><div><p class="MsoNormal"> From: "Liam - Test" <<a href="mailto:sip%3A12345678@9.8.7.6" target="_blank">sip:12345678@9.8.7.6</a>>;tag=ryFQea59e6Upe</p>
</div><div><p class="MsoNormal"> To: <<a href="http://sip:0800000000@1.2.3.4:5060" target="_blank">sip:0800000000@1.2.3.4:5060</a>></p></div><div><p class="MsoNormal"> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43</p>
</div><div><p class="MsoNormal"> CSeq: 12805602 INVITE</p></div><div><p class="MsoNormal"> Contact: <<a href="http://sip:12345678@9.8.7.6:5060" target="_blank">sip:12345678@9.8.7.6:5060</a>></p></div><div><p class="MsoNormal">
User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported</p></div><div><p class="MsoNormal"> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY</p></div><div><p class="MsoNormal"> Supported: timer, precondition, path, replaces</p>
</div><div><p class="MsoNormal"> Allow-Events: talk, hold, refer</p></div><div><p class="MsoNormal"> Content-Type: application/sdp</p></div><div><p class="MsoNormal"> Content-Disposition: session</p></div><div><p class="MsoNormal">
Content-Length: 189</p></div><div><p class="MsoNormal"> X-FS-Support: update_display</p></div><div><p class="MsoNormal"> Remote-Party-ID: "Liam - Test" <<a href="mailto:sip%3A12345678@9.8.7.6" target="_blank">sip:12345678@9.8.7.6</a>>;party=calling;screen=yes;privacy=off</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal"> v=0</p></div><div><p class="MsoNormal"> o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6</p></div><div><p class="MsoNormal"> s=FreeSWITCH</p></div>
<div><p class="MsoNormal"> c=IN IP4 9.8.7.6</p></div><div><p class="MsoNormal"> t=0 0</p></div><div><p class="MsoNormal"> m=audio 18646 RTP/AVP 0 8 101 13</p></div><div><p class="MsoNormal"> a=rtpmap:101 telephone-event/8000</p>
</div><div><p class="MsoNormal"> a=fmtp:101 0-16</p></div><div><p class="MsoNormal"> ------------------------------------------------------------------------</p></div></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">
</p></div><div><p class="MsoNormal">Thanks</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal"><span style="color:#888888">Liam</span></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_______________________________________________<br>
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