Who is answering the call? Are you sure they're not choosing a different codec from the ones offered in the INVITE?<br><br>-Steve<br><br><br><div class="gmail_quote">On 23 May 2011 16:52, Sean Eichhorn <span dir="ltr"><<a href="mailto:seichhorn@gci.com">seichhorn@gci.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div link="blue" vlink="purple" style="word-wrap:break-word" lang="EN-US">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Yes, I understand that this is the Cisco proprietary codec. The
problem is that between two Cisco systems the SIP endpoints will not use this
protocol if it is not sent in the SDP. The *<b>original</b>* SDP does contain
the codec, but when the call is answered, another SDP is sent. This SDP is
altered by Freeswitch and all codecs other than the selected codec, NTE, and comfort
noise are removed.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">It might not be Freeswitch doing it. I’m digging into the
source and it looks like it might be the Sofia-SIP library.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">The funny thing is that it works in one direction, but not the
other.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">FS subscriber -> CIDR defined static SIP endpoint (WORKS!)</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">CIDR defined SIP endpoint -> FS Subscriber (Doesn’t work)</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">It seems that one of the major differences is that the
subscriber answers with a 200 OK, whereas the CIDR endpoint answers with a 183.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">I’m currently searching to see if there is a way to change the
response type, but so far no luck.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">Sean Eichhorn</span><span style="color:#1F497D"></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">General Communications Inc.</span><span style="color:#1F497D"></span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">IP Telephony Engineer</span><span style="color:#1F497D"></span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">(907) 868-6902</span><span style="color:#1F497D"></span></p>
</div>
<div>
<p class="MsoNormal"><span style="color:#1F497D"> </span></p>
</div>
</div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">-No telephony products were harmed in the making of the message-</span><span style="font-size:11.0pt;color:#1F497D"></span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> David Ponzone
[mailto:<a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a>] <br>
<b>Sent:</b> Friday, May 20, 2011 01:03<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] rtpmap line missing on answer</span></p>
</div>
</div><div><div></div><div class="h5">
<p class="MsoNormal"> </p>
<p class="MsoNormal">Sean,e <br></p>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">X-NSE is the Clear Channel codec from Cisco.</p>
</div>
<div>
<p class="MsoNormal">I suspect FreeSWITCH does not support it, and as such, it
can't be offered to leg B.</p>
</div>
<div>
<p class="MsoNormal">Perhaps there is a way to enable in outbound codecs as
bypass, but I really doubt so.</p>
</div>
<div>
<p class="MsoNormal">Though, you could try to enable late-negotiation where
client and gateway will negotiate together.</p>
</div>
<div>
<p class="MsoNormal"> </p>
<div>
<div>
<div>
<p class="MsoNormal"><span style="font-size:10.5pt">David Ponzone </span><span><span style="font-size:9.0pt">Direction
Technique</span></span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<p class="MsoNormal"><span><span style="font-size:10.0pt">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<p class="MsoNormal"><span><span style="font-size:10.0pt">tel: 01
74 03 18 97</span></span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<p class="MsoNormal"><span><span style="font-size:10.0pt">gsm: 06 66 98 76 34</span></span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:10.5pt;color:black"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:10.5pt">Service Client<span> </span></span><span style="font-size:10.5pt">IP</span><span style="font-size:10.5pt">eva</span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<div>
<p class="MsoNormal"><span><span style="font-size:10.0pt">tel:
0811 46 26 26</span></span><span style="font-size:10.5pt;color:black"></span></p>
</div>
<div>
<div>
<p class="MsoNormal"><u><span style="font-size:7.5pt;color:#0022F3"><a>www.ipeva.fr</a></span></u><span style="font-size:7.5pt;color:#656895">
- </span><u><span style="font-size:7.5pt;color:#0022F3"><a>www.ipeva-studio.com</a></span></u><span style="font-size:7.5pt;color:#0022F3"></span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:7.5pt;color:#0022F3"> </span></p>
</div>
<div>
<p class="MsoNormal" style="text-align:justify"><i><span style="font-size:7.5pt;color:silver">Ce message et toutes les pièces
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<p class="MsoNormal" style="text-align:justify"><u><span style="font-size:7.5pt;color:#0022F3"><span style="text-decoration:none"> </span></span></u></p>
</div>
</div>
</div>
</div>
<p class="MsoNormal"><span style="font-size:10.5pt;color:black"><br>
<br>
</span></p>
</div>
<p class="MsoNormal"> </p>
<div>
<div>
<p class="MsoNormal">Le 20/05/2011 à 20:54, Sean Eichhorn a écrit :</p>
</div>
<p class="MsoNormal"><br>
<br>
</p>
<div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">I
have a freeswitch system that (for the most part) works flawlessly for
me. However, I need to have an additional rtpmap line in the SDP when a
call is answered.</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">My
client sends the following SDP upon answering the call :</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
m=audio 17214 RTP/AVP 0 19 101 100</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
c=IN IP4 192.168.98.79</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:0 PCMU/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:19 CN/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:101 telephone-event/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=fmtp:101 0-16</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:100 X-NSE/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=fmtp:100 192-194</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">Freeswitch
forwards the following SDP to the external gateway :</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
m=audio 20654 RTP/AVP 0 19 101</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
c=IN IP4 66.223.187.208</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:0 PCMU/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:19 CN/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=rtpmap:101 telephone-event/8000</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">
a=fmtp:101 0-16</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">The
other information appears fine. Everything works, but my client-side is
not receiving the a=rtpmap:100 line.</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">This
issue appears regardless of whether or not I’m in proxy mode, bypass mode, or
neither.</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">Using
"sip_append_audio_sdp” has no effect on the answering SDP, only the
initial offer.</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">Any
ideas?</span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt">Thanks
in advance,</span></p>
</div>
<p class="MsoNormal"><span style="font-size:13.5pt">_______________________________________________<br>
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</div>
</div>
<p class="MsoNormal"> </p>
</div>
</div></div></div>
</div>
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