WebRTC client

cowwoc cowwoc at bbs.darktech.org
Wed Dec 5 08:26:51 PST 2012

Hi Nicolas,

On 05/12/2012 11:03 AM, Nicolas Dufresne [via GStreamer-devel] wrote:
> Good point, I forgot about SRTP. There is some effort that already
> started [1].

     I believe that effort has stalled. I exchanged emails with Olivier 
two days ago and he wrote: "You are correct that SRTP is a missing 
block, I'm not sure which thing to use to implement it. I'm not sure 
which SRTP variant will be adopted by the rtcweb workgroup, if it is 
DTLS-SRTP, then maybe we should go with something like openssl or 
gnutls, but I haven't investigated that fully. "

> For iSAC and iLBC I don't know exactly what it is. From
> quick googling they seems to be speech base audio codecs, which in this
> case shall be provided by the speex element. Payloader/Depayloader might
> be missing, but writing those is fairly simple. Chrome negotiate codecs,
> so I would not worry too much about that part.

     Hmm, I'm not sure this would work (who says WebRTC is supposed to 
support Speex?) but in any case I've got some excellent news. I just 
read at http://www.webrtc.org/faq-recent-topics that (at least on paper) 
Chrome 24 will introduce Opus support. So I think the only missing 
pieces here are:

 1. SRTP support
 2. PeerConnection implementation.

     http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-05 indicates 
WebRTC uses the DTLS-SRTP variant.
     I believe http://tools.ietf.org/html/draft-uberti-rtcweb-jsep-02 
describes how to implement PeerConnection.

     Any idea how much would it would be to add these to GStreamer?


> best regards,
> Nicolas
> [1] https://bugzilla.gnome.org/show_bug.cgi?id=632206
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