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<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US">Dear Steve.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US">Asterisk is a B2BUA. It doesn’t just route SIP messages like Kamailio does. Most of the time, it answers the call, sets up a new call to the other party and then bridges the two calls,
making two separate call IDs (depending on your configuration).<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US">Additional comments in blue inline below:<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<ul style="margin-top:0cm" type="disc">
<li class="MsoNormal" style="mso-list:l0 level1 lfo3"><b><span lang="EN-US">INVITE</span></b><span lang="EN-US"> from Caller to Kamailio (this is where the Caller is Registered) offering VP8<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Kamailio will relay the INVITE to Asterisk<o:p></o:p></span></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"><span lang="EN-US"> Asterisk probably sends 183/200 back to callee, finalizing this codec negotiation. VP8 is chosen for this call leg.<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Asterisk will
<b>INVITE</b> Called <o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Called sends ‘200 OK’ to Asterisk requesting H264<o:p></o:p></span></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"><span lang="EN-US"> yes<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Asterisk Sends ‘200 OK’ back to Kamailio to be relayed to Caller accepting VP8<o:p></o:p></span></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"><span lang="EN-US"> No. I assume that the callee didn’t accept VP8 because it doesn’t support it. It instead chose the highest priority codec it supported among the codecs that Asterisk advertised.<o:p></o:p></span></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"><span lang="EN-US"> Asterisk accepts this codec because it supports it. There is no test checking which codec was chosen on the caller side and there is no renegotiation. Hence, video is broken
forever.<o:p></o:p></span></li></ul>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US">Again, if you could show us the details (like attaching the SDPs that I asked for previously), then we wouldn’t have to guess what’s happening. We could tell you for sure.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="background:white"><span lang="EN-US" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#4B5669">Med vennlig hilsen</span><span lang="EN-US" style="font-size:11.5pt;color:#212121"><o:p></o:p></span></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#4B5669">Pan B. Christensen<o:p></o:p></span></p>
</div>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> sr-users <sr-users-bounces@lists.kamailio.org>
<b>On Behalf Of </b>Wilkins, Steve<br>
<b>Sent:</b> fredag 8. juni 2018 22:27<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org><br>
<b>Subject:</b> Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span lang="EN-US">Hi Pan,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">If the Registered caller is a WebRTC client (Caller) whose preferred coded is VP8, calls a Phone (Called) whose preferred Video is H264, I should see<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<ul style="margin-top:0cm" type="disc">
<li class="MsoNormal" style="mso-list:l0 level1 lfo3"><b><span lang="EN-US">INVITE</span></b><span lang="EN-US"> from Caller to Kamailio (this is where the Caller is Registered) offering VP8<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Kamailio will relay the INVITE to Asterisk<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Asterisk will
<b>INVITE</b> Called <o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Called sends ‘200 OK’ to Asterisk requesting H264<o:p></o:p></span></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3"><span lang="EN-US">Asterisk Sends ‘200 OK’ back to Kamailio to be relayed to Caller accepting VP8<o:p></o:p></span></li></ul>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Is this correct? Then I think one of the issues is that there is no fmtp line in the VP8. The only codecs that have fmtp lines is for the H264 codecs.
<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thank you,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">-Steve<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> sr-users [<a href="mailto:sr-users-bounces@lists.kamailio.org">mailto:sr-users-bounces@lists.kamailio.org</a>]
<b>On Behalf Of </b>Pan Christensen<br>
<b>Sent:</b> Friday, June 8, 2018 11:06 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">That’s not strange. The soft phone probably doesn’t advertise support for VP8, so it’s not chosen.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk accepts it. Asterisk then finds out that the soft phone doesn’t support VP8 and negotiates a different codec with that client.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the order they are sent), we can tell you exactly what happens. Failing that, I’d say that the culprit is Asterisk, which probably negotiates
two different codecs without the ability to transcode.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span lang="EN-US" style="font-size:9.0pt;font-family:"Helvetica",sans-serif;color:black">With kind regards<br>
<b>Pan B. Christensen</b><br>
Developer<br>
Phonect AS </span><span lang="EN-US"><o:p></o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> sr-users <<a href="mailto:sr-users-bounces@lists.kamailio.org">sr-users-bounces@lists.kamailio.org</a>>
<b>On Behalf Of </b>Wilkins, Steve<br>
<b>Sent:</b> fredag 8. juni 2018 15:09<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></span></p>
</div>
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<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264). What is strange is that if it is the other way around and the Soft-Phone calls the WebRTC client, it works.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thank you<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> sr-users [<a href="mailto:sr-users-bounces@lists.kamailio.org">mailto:sr-users-bounces@lists.kamailio.org</a>]
<b>On Behalf Of </b>Pan Christensen<br>
<b>Sent:</b> Friday, June 8, 2018 9:00 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Hello Steve.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Does Asterisk negotiate different codecs with each client? If so, it needs to transcode, which I believe is currently not supported for video. What does Asterisk send back to device A?<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<div>
<p class="MsoNormal"><span lang="EN-US" style="font-size:9.0pt;font-family:"Helvetica",sans-serif;color:black">With kind regards<br>
<b>Pan B. Christensen</b><br>
Developer<br>
Phonect AS </span><span lang="EN-US"><o:p></o:p></span></p>
</div>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> sr-users <<a href="mailto:sr-users-bounces@lists.kamailio.org">sr-users-bounces@lists.kamailio.org</a>>
<b>On Behalf Of </b>Wilkins, Steve<br>
<b>Sent:</b> fredag 8. juni 2018 14:03<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Hi All,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses incorrect Video RTP Payload Type when sending Video packets.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I have a situation where I make a call from Device A to Device B and Device A is Registered in Kamailio. When Device A Calls Device B, Kamailio sends an ‘INVITE’ to Asterisk, Asterisk then ‘INVITES’ Device B. I get
two-way Audio, the call stays connected, however, when Video packets are sent to Device B, the RTP Payload Type is incorrect. The port is correct, but just not the Payload Type.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Here is where I think Kamailio is involved. In the first Invite from Kamailio to Asterisk, one of the offered Video codecs is ‘100 H264’; interesting enough, Device B wants to use ‘115 H264’ and when Asterisk sends out
Video packets, it is using ‘100’ instead of ‘115’, and of course I have no Video. I don’t know if this is just a coincidence but it sure seems like that is where the issue may lie.
<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Has anyone ever seen this behavior? The Asterisk teams does not think it’s an Asterisk issue.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thank you,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">-Steve<o:p></o:p></span></p>
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