I just noticed the value of media.navigator.video.h264.level: media.navigator.video.h264.level = 31
Does this means that the only level supported is 3.1? Thanks, On Thursday, September 4, 2014 2:12:59 PM UTC-6, [email protected] wrote: > Hi, > > > > In the past days I've been trying to display an H.264 stream generated by > Gstreamer 0.10 in Firefox Nightly (35.0a1), using WebRTC. I'm in Ubuntu 12.04. > > > > I've enabled the "media.peerconnection.video.h264_enabled" flag, and from > there on I have tried several paths. > > > > 1) First, tried the H.264 SDP demo from the WebRTC book [1] and I was able to > run it using two Firefox Nightly's in different computers and successfully > make a call between them. From this demo, I think the important parameters in > the SDP offer were: > > > > a=rtpmap:126 H264/90000 > > a=fmtp:126 profile-level-id=42e00c;packetization-mode=1 > > > > Also based in the "H.264 as Mandatory to Implement Video Codec for WebRTC" > draft [2]. > > > > 2) Janus WebRTC Gateway > > > > The Janus Gateway has a demo [3] where a gstreamer pipeline based on Opus/VP8 > is displayed in the browser. My goal was to modify this demo in order to use > an H.264 gstreamer pipeline as the stream generator. In concept, Janus > doesn't mind about codecs but just negotiation. So I've modified that demo in > Janus to change the SDP offer to H.264, setting the rtpmap and fmtp lines > parameters as in 1) above, and running a gstreamer pipeline that (I think) > would comply to those parameters (especially payload type and > profile-level-id): > > > > gst-launch videotestsrc ! video/x-raw-rgb,width=320,height=240,framerate=15/1 > ! videoscale ! videorate ! ffmpegcolorspace ! timeoverlay ! x264enc ! > rtph264pay pt=126 profile-level-id=42e00c ! udpsink host=127.0.0.1 port=5008 > > > > I've been able to start the demo (the negotiation seems to succeed), but > Firefox doesn't display the video. The details are in [4] from Janus' > discussion forum. > > > > 3) Recenlty, someone had a similar requirement in the stack overflow question > "WebRTC and gstreamer on linux device" [5], but was thinking on using a > node.js server instead. I haven't explored that path yet. > > > > So, I think in summary, I have some questions that I will appreciate if you > help me understand. > > > > Q1) What is the state of H.264 WebRTC support in the latest Firefox Nightly - > what are the profiles, levels, supported? > > Q2) Has anyone tried H.264 WebRTC streaming that is not between two browsers? > (I.e. my setup of Gstreamer --> WebRTC gateway <--> WebRTC browser) > > Q3) Any hint on why my test using Janus Gateway is not working? (See [4] for > details) > > > > And, overall, I have had a hard time understanding how can I know what > profile-level-id I must set in the SDP fmtp parameters. Any help on this > direction is also welcomed. > > > > Thanks much for your help. > > > > [1] http://webrtcbook.com/sdp-h264.html > > [2] http://tools.ietf.org/html/draft-burman-rtcweb-h264-proposal-04#section-9 > > [3] https://janus.conf.meetecho.com/streamingtest.html > > [4] https://groups.google.com/forum/#!topic/meetecho-janus/XjjboCTY0Xc > > [5] > http://stackoverflow.com/questions/25463064/webrtc-and-gstreamer-on-linux-device > > > > Regards, > > JP _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
