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Hi,
What is the status of getting GStreamer to act as a WebRTC client? I saw some old posts on the Wiki but it's not clear what pieces are needed to get this to work and what their current status is. Thank you, Gili |
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Le lundi 03 décembre 2012 à 11:13 -0800, cowwoc a écrit :
> Hi, > > What is the status of getting GStreamer to act as a WebRTC client? I saw > some old posts on the Wiki but it's not clear what pieces are needed to get > this to work and what their current status is. There is a bug on bugs.webkit.org [1] to track the effort on WebKitGTK side (see dependencies of this bug). GStreamer combined with Farstream shall provide all you need to implement such a client. regards, Nicolas [1] https://bugs.webkit.org/show_bug.cgi?id=87514 _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Nicolas,
GStreamer supports VP8 and Opus (on paper at least) but the current implementation of WebRTC (found in Chrome 23) requires SRTP, iSAC and iLBC which GStreamer does *not* support (not even Farstream to my knowledge). What do we do about that? References: http://www.webrtc.org/reference/architecture and http://code.google.com/p/chromium/issues/detail?id=104241 Gili On 05/12/2012 10:16 AM, Nicolas Dufresne [via GStreamer-devel] wrote: Le lundi 03 décembre 2012 à 11:13 -0800, cowwoc a écrit : ... [show rest of quote]
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Le mercredi 05 décembre 2012 à 07:36 -0800, cowwoc a écrit :
> GStreamer supports VP8 and Opus (on paper at least) but the > current implementation of WebRTC (found in Chrome 23) requires SRTP, > iSAC and iLBC which GStreamer does *not* support (not even Farstream > to my knowledge). What do we do about that? Good point, I forgot about SRTP. There is some effort that already started [1]. 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. best regards, Nicolas [1] https://bugzilla.gnome.org/show_bug.cgi?id=632206 _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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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 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 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:
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? Thanks, Gili
... [show rest of quote]
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Hi,
There is a patch on bugzilla that adds a ilbc plug in, although this is based on the reference implementation and Google has recently opened a better implementation. That said, I doubt that ilbc and isac will be in the final webrtc as they are little used outside of GIPS (now owned by Google) and Opus is better in every way. -- Sent from my Android phone with K-9 Mail. Please excuse my brevity. _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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In reply to this post by cowwoc
Le mercredi 05 décembre 2012 à 08:26 -0800, cowwoc a écrit :
> openssl or gnutls, but I haven't investigated that fully. " These days, GLib propose an abstracted SSL API, this shall be used imho. Nicolas _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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In reply to this post by cowwoc
Le mercredi 05 décembre 2012 à 08:26 -0800, cowwoc a écrit :
> PeerConnection implementation. I believe this is JavaScript specific no ? _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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I don't think so. Users pass SDP and a list of ICE candidates into a PeerConnection implementation and it establishes a P2P connection with the remote peer. Gili On 05/12/2012 1:19 PM, Nicolas Dufresne [via GStreamer-devel] wrote: Le mercredi 05 décembre 2012 à 08:26 -0800, cowwoc a écrit : |
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In reply to this post by Nicolas Dufresne
That would be good and all only if it exposes the DTLS bits and I don't think it does..
Nicolas Dufresne <[hidden email]> wrote: >Le mercredi 05 décembre 2012 à 08:26 -0800, cowwoc a écrit : >> openssl or gnutls, but I haven't investigated that fully. " > >These days, GLib propose an abstracted SSL API, this shall be used >imho. > >Nicolas > >_______________________________________________ >gstreamer-devel mailing list >[hidden email] >http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel -- Sent from my Android phone with K-9 Mail. Please excuse my brevity. _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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In reply to this post by Nicolas Dufresne
Hi there,
You name sounds french, I am trying to establish contacts with WebRTC developpers (preferably french speaking ones) in view of developping a new web application. Can you get in contact with me (in french or english) in you can help or if you can recommend one of your friends. Thanks in advance Frédéric Eppendahl frederic@eppendahl.com |
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In reply to this post by Olivier Crête-3
You name sounds french,
I am trying to establish contacts with WebRTC developpers (preferably french speaking ones) in view of developping a new web application. Can you get in contact with me (in french or english) if you can help or if you can recommend one of your friends. Thanks in advance Frédéric Eppendahl frederic@eppendahl.com |
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In reply to this post by cowwoc
Le mercredi 05 décembre 2012 à 12:43 -0800, cowwoc a écrit :
> I don't think so. Users pass SDP and a list of ICE candidates into > a PeerConnection implementation and it establishes a P2P connection > with the remote peer. Yes, I mean it's not GStreamer specific, it's all about the glue code between JavaScript and what you are basing your work on. At the least this should be fully supported by libnice/farstream atm. Nicolas _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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