[OpenSIPS-Users] h264 webrtc and opensips

[Esty, Ryan]

Razvan,

Thanks for getting back to me. I was afraid of this as I didn't see any options in rtpengine that supported video codecs either. We are trying to upgrade some of our devices to support packetization-mode 0 and 1.

Ryan Esty
Senior Software Engineer
NEC Enterprise Communication Technologies (Cheshire)
203-718-6268

-----Original Message-----
From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Tuesday, April 3, 2018 1:19 PM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] h264 webrtc and opensips

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
http://www.opensips-solutions.com

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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> Users mailing list
> Users at lists.opensips.org
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