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<p class="MsoNormal">Pan,<o:p></o:p></p>
<p class="MsoNormal">I did verify the ‘200 OK’ is making it to the softphone because I see the ‘ACK’ to it. So that isn’t the issue.<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> Saturday, June 9, 2018 6:20 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">I have to admit that I don’t have much experience with video calls, but as far as I can tell VP8 is negotiated with WebRTC client. Nothing is negotiated with softphone.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">WebRTC client offers:<o:p></o:p></p>
<p class="MsoNormal">Media Description, name and address (m): video 55185 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Asterisk offers to softphone:<o:p></o:p></p>
<p class="MsoNormal">Media Description, name and address (m): video 15112 RTP/AVP 99<o:p></o:p></p>
<p class="MsoNormal">Why is it advertising only 99?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Softphone answers with:<o:p></o:p></p>
<p class="MsoNormal">Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31<o:p></o:p></p>
<p class="MsoNormal">No match with offer, video is not negotiated for this call leg.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Asterisk answers WebRTC with:<o:p></o:p></p>
<p class="MsoNormal">Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100<o:p></o:p></p>
<p class="MsoNormal">96 is chosen for this call leg.<o:p></o:p></p>
<p class="MsoNormal"><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> lørdag 9. juni 2018 03:59<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 again Pan and Thank You for looking at this!! Here is the pcap for the call that is not showing video on either side of the call. As I mentioned I do get two way Audio, the call stays connected, but just no Video. The Video ports
look to be correct. Video packets appear to go out (at least packets are going out over the Video ports). The screen on the Caller (A WebRTC Client) and the Called (A Provider Phone) are both blank. The WebRTC client is Registered in Kamailio<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Note: when I sanitized the file using tcprewrite an extra RTP packet appeared (Unknown RTP packet version 0), this was not here in the original.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I use Kamailio 5.2.0 and Asterisk 15.3<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you again!, <o:p></o:p></p>
<p class="MsoNormal">-Steve<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 5:12 PM<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">Dear Steve.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Additional comments in blue inline below:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<ul style="margin-top:0in" type="disc">
<li class="MsoNormal" style="mso-list:l0 level1 lfo3"><b>INVITE</b> from Caller to Kamailio (this is where the Caller is Registered) offering VP8<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Kamailio will relay the INVITE to Asterisk<o:p></o:p></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"> Asterisk probably sends 183/200 back to callee, finalizing this codec negotiation. VP8 is chosen for this call leg.<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Asterisk will <b>INVITE</b> Called
<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Called sends ‘200 OK’ to Asterisk requesting H264<o:p></o:p></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"> yes<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Asterisk Sends ‘200 OK’ back to Kamailio to be relayed to Caller accepting VP8<o:p></o:p></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"> 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></li><li class="MsoNormal" style="color:#0070C0;mso-list:l0 level1 lfo3"> 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></li></ul>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#4B5669">Med vennlig hilsen</span><span style="font-size:11.5pt;color:#212121"><o:p></o:p></span></p>
<p class="MsoNormal" style="background:white"><span lang="NO-BOK" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#4B5669">Pan B. Christensen<o:p></o:p></span></p>
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<p class="MsoNormal"><span lang="NO-BOK"><o:p> </o:p></span></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 22:27<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"><span lang="NO-BOK"><o:p> </o:p></span></p>
<p class="MsoNormal">Hi Pan,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></p>
<p class="MsoNormal"><o:p> </o:p></p>
<ul style="margin-top:0in" type="disc">
<li class="MsoNormal" style="mso-list:l0 level1 lfo3"><b>INVITE</b> from Caller to Kamailio (this is where the Caller is Registered) offering VP8<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Kamailio will relay the INVITE to Asterisk<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Asterisk will <b>INVITE</b> Called
<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Called sends ‘200 OK’ to Asterisk requesting H264<o:p></o:p></li><li class="MsoNormal" style="mso-list:l0 level1 lfo3">Asterisk Sends ‘200 OK’ back to Kamailio to be relayed to Caller accepting VP8<o:p></o:p></li></ul>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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></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>
<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 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></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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