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<p class="MsoNormal">Sorry Pan, but one other thing I noticed. When Asterisk send the 200 OK, Kamailio does see it, but I don’t see where it is relaying it to the WebRTC client that made the call. However, it must, otherwise the call would disconnect in 30
seconds and also, I do have two-way Audio.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks again for you input on this. It’s seems a bit confusing as to where an issue lies.<o:p></o:p></p>
<p class="MsoNormal"><a name="_MailEndCompose"><o:p> </o:p></a></p>
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<p class="MsoNormal"><b>From:</b> sr-users [mailto:sr-users-bounces@lists.kamailio.org]
<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 <sr-users@lists.kamailio.org><br>
<b>Subject:</b> Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">That’s not strange. The soft phone probably doesn’t advertise support for VP8, so it’s not chosen.<o:p></o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span 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><o:p></o:p></p>
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<p class="MsoNormal"><b>From:</b> 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></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b>From:</b> 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></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hello Steve.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span 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><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b>From:</b> 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></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hi All,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Has anyone ever seen this behavior? The Asterisk teams does not think it’s an Asterisk issue.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you,<o:p></o:p></p>
<p class="MsoNormal">-Steve<o:p></o:p></p>
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