[OpenSIPS-Users] h264 webrtc and opensips

Esty, Ryan ryan.esty at necect.com
Fri Mar 23 11:01:25 EDT 2018

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

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