[OpenSIPS-Users] h264 webrtc and opensips

Răzvan Crainea razvan at opensips.org
Tue Apr 3 13:18:38 EDT 2018

Hi, Ryan!

I don't have that much experience with H.264, but my first instinct was 
to look into the rtpengine packetization feature. But unfortunately 
rtpengine does not support H.264 codecs, so I doubt this can help. But 
perhaps you could look into different transcoding solutions that do 
support H.264 transcoding.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer

On 03/23/2018 05:01 PM, Esty, Ryan wrote:
> 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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