[OpenSIPS-Users] h264 webrtc and opensips

[Esty, Ryan]

Hi list,

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 (https://tools.ietf.org/html/rfc7742).

Ryan Esty

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20180323/136dac0e/attachment.html>