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<p class="MsoNormal">Hi list,<o:p></o:p></p>
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<p class="MsoNormal">This might not be the correct list for this but maybe someone might be able to point me in the correct direction. I’m trying to use opensips as a webrtc gateway. It mostly works I’m able to call a legacy sip phone connected to my SIP server.
The reason why it only mostly works is I have a problem with the h264 codec. None of my legacy devices know what to do with packetization-mode=1, well this is my assumption. Has anyone else had a similar issue and can point me to some further information?
A lot of people said to just set packetization-mode to 0 but I thought the webrtc video draft said this was mandatory (<a href="https://tools.ietf.org/html/rfc7742">https://tools.ietf.org/html/rfc7742</a>).<o:p></o:p></p>
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<p class="MsoNormal">Ryan Esty<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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