using webrtcbin for regular RTP and SRTP flows

Daniel Pocock daniel at
Mon Apr 12 21:39:50 UTC 2021

Hi Olivier,

I had a quick look over the Farstream page

The overall solution (mentioned in the other thread about rtpbin) needs
to accept connections from either regular clients or WebRTC clients

Can Farstream be mixed with any arbitrary Gstreamer pipelines, for
example, if a call ends up with regular RTP on one side and webrtcbin on
the other end of the pipeline?

Incidentally, we also have a telepathy module in the reSIProcate tree,
it would be interesting to have it interoperate from Telepathy to a
browser (JsSIP) using webrtcbin:



On 12/04/2021 22:25, Olivier Crête wrote:
> Hi,
> The answer is no... You may want to look at Farstream, which we design
> for traditional SIP/XMPP.
> Olivier
> On Mon, 2021-04-12 at 22:10 +0200, Daniel Pocock wrote:
>> Is it possible to give the webrtcbin element an SDP offer for a
>> traditional SIP call without full WebRTC?
>> For example, an SDP with:
>> - no ICE candidates, just regular connection,
>> - no TURN,
>> - no DTLS-SRTP, maybe regular SDES or straight RTP
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at

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