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<p class="MsoNormal">Hello,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span lang="EN-GB">AFAIK there is no support in rtpengine for video codecs yet, e.g., refer to this discussion:</span><span lang="EN-GB">
</span><span lang="EN-GB"><a href="https://groups.google.com/g/rtpengine/c/Tf3us2oBW3w">https://groups.google.com/g/rtpengine/c/Tf3us2oBW3w</a><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">You can of course route the calls with video differently with Kamailio, for example with sdpops or other detection mechanism. And then using one of the mentioned other PBX projects to process the call.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">I would probably try to handle the video call media completely on the PBX then, and not try to mix and match somehow it on the different components. Especially as its only a low percentage of the total calls.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">Cheers,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">Henning<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span lang="EN-GB">-- <o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">Henning Westerholt – </span><a href="https://skalatan.de/blog/"><span lang="EN-GB">https://skalatan.de/blog/</span></a><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB">Kamailio services – </span><a href="https://gilawa.com/"><span lang="EN-GB">https://gilawa.com</span></a><span lang="EN-GB"><o:p></o:p></span></p>
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<p class="MsoNormal"><span lang="EN-GB"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><b><span style="mso-fareast-language:DE">From:</span></b><span style="mso-fareast-language:DE"> sr-users <sr-users-bounces@lists.kamailio.org>
<b>On Behalf Of </b>Rüdiger, Christoph<br>
<b>Sent:</b> Friday, May 13, 2022 6:54 PM<br>
<b>To:</b> sr-users@lists.kamailio.org<br>
<b>Subject:</b> [SR-Users] Split audio and video into different calls<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<p class="MsoNormal" style="margin-left:35.4pt">Hi everyone,<o:p></o:p></p>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">we are running a call center with really good to use hardphones and audio only up to this point. Because of changing regulation, we soon have to deal with video calls as well, but only for a
fraction of the calls (below 1%).<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">Instead of switching everyone to clumsy softphones for the rare case of a video call, I’m exploring the ability to branch an incoming call into a pure audio call to the hardphone and optionally
a pure video call to the softphone.<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">Has anyone done this in the past and can share some experiences/ideas on<o:p></o:p></span></p>
<p class="MsoListParagraph" style="margin-left:71.4pt;text-indent:-18.0pt;mso-list:l1 level1 lfo3">
<![if !supportLists]><span lang="EN-US"><span style="mso-list:Ignore">a)<span style="font:7.0pt "Times New Roman"">
</span></span></span><![endif]><span lang="EN-US">Can it be done purely with Kamailio and rtpengine? I don’t have a clear picture of how to do it purely with Kamailio on the SIP side, because we end up with two independent legs (to the hardphone and the softphone).
Or do I need to deploy an Asterisk, Yate, freeswitch, etc. and doing it as a 3-way conference where one party gets both RTP streams (the trunk) and the other two parties only get one RTP stream (either audio or video)?<o:p></o:p></span></p>
<p class="MsoListParagraph" style="margin-left:71.4pt;text-indent:-18.0pt;mso-list:l1 level1 lfo3">
<![if !supportLists]><span lang="EN-US"><span style="mso-list:Ignore">b)<span style="font:7.0pt "Times New Roman"">
</span></span></span><![endif]><span lang="EN-US">Was it worth the effort or a steady candidate for troubles?<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">I’m looking forward to your thoughts, especially on question b).<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">Have a nice weekend,<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">Christoph<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-US">--- <o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black;mso-fareast-language:DE">Christoph Rüdiger<o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt">
<span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black;mso-fareast-language:DE">Digital Operations Architect<br>
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</span><span style="font-size:8.0pt;font-family:"Arial",sans-serif;color:black;mso-fareast-language:DE"><br>
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