Difference between revisions of "GstWebRTC - PubNub Audio + Video Examples - x86"

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In this example we use rrwebrtcsink to send a video stream and an audio stream, then rrwebrtcsrc receive the video and audio streams.
 
In this example we use rrwebrtcsink to send a video stream and an audio stream, then rrwebrtcsrc receive the video and audio streams.
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<pre style="background-color:yellow">
 
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It seems that browsers do not get along with x264 because of SEI NAL units sent with the stream. As a workaround, we set key-int-max=1 and avoid the SEI insertions.
 
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</pre>
 
  
 
==== Send Pipeline ====
 
==== Send Pipeline ====
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PEER_CHANNEL=123peer
 
PEER_CHANNEL=123peer
 
gst-launch-1.0 rrwebrtcsink rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
 
gst-launch-1.0 rrwebrtcsink rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
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videotestsrc is-live=true ! x264enc aud=false key-int-max=1 tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video \
+
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video \
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio
 
</syntaxhighlight>
 
</syntaxhighlight>
Line 57: Line 53:
 
PEER_CHANNEL=123peer
 
PEER_CHANNEL=123peer
 
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
 
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
−
videotestsrc is-live=true ! x264enc aud=false key-int-max=1 tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
+
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
 
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
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PEER_CHANNEL=123
 
PEER_CHANNEL=123
 
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
 
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
−
videotestsrc is-live=true ! x264enc aud=false key-int-max=1 tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
+
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
 
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \

Revision as of 20:29, 9 January 2019

Error something wrong.jpg Problems running the pipelines shown on this page?
Please see our GStreamer Debugging guide for help.

←

PubNub Video Examples

⌂
Home 		→

PubNub Web Pages


This page links to the GstWebRTC audio and video examples using PubNub.


H264+Opus

Unidirectional elements

Example

In this example we use rrwebrtcsink to send a video stream and an audio stream, then rrwebrtcsrc receive the video and audio streams.

Send Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcsink rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio

Receive Pipeline

USER_CHANNEL=123peer
PEER_CHANNEL=123
gst-launch-1.0 rrwebrtcsrc rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
web.video ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
web.audio ! rtpopusdepay ! opusdec ! autoaudiosink

Bidirectional elements

Example

In this example we use two rrwebrtcbins, each send a video stream and an audio stream, and receives each other video and audio streams.

Send-Receive Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink

Send-Receive Pipeline

USER_CHANNEL=123peer
PEER_CHANNEL=123
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! x264enc tune=zerolatency intra-refresh=true ! "video/x-h264,profile=constrained-baseline,level=(string)3.1" ! rtph264pay ! web.video_sink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
web.video_src ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink

VP8+Opus

Unidirectional elements

Example

In this example we use rrwebrtcsink to send a video stream and an audio stream, then rrwebrtcsrc receive the video and audio streams.

Send Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcsink rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio

Receive Pipeline

USER_CHANNEL=123peer
PEER_CHANNEL=123
gst-launch-1.0 rrwebrtcsrc rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
web.video ! rtpvp8depay ! avdec_vp8 ! autovideosink \
web.audio ! rtpopusdepay ! opusdec ! autoaudiosink

Bidirectional elements

Example

In this example we use two rrwebrtcbins, each send a video stream and an audio stream, then receives each other video and audio streams.

Send-Receive Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
web.video_src ! rtpvp8depay ! avdec_vp8 ! autovideosink \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink

Send-Receive Pipeline

USER_CHANNEL=123peer
PEER_CHANNEL=123
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
web.video_src ! rtpvp8depay ! avdec_vp8 ! autovideosink \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink


←

PubNub Video Examples

⌂
Home 		→

PubNub Web Pages

  1. WebRTC Fundamentals
  2. GstWebRTC Basics
    1. Signaling
      1. Predefined Signaling
      2. Custom Signaling
      3. API Reference
      4. Advance Settings
    2. Plugin Elements
      1. GstWebRTCBin
      2. GstWebRTCBin Properties
      3. GstWebRTCBin Features
        1. Data Channel
  3. Evaluating GstWebRTC
  4. Getting the code
  5. Building GstWebRTC
    1. x86 Platforms
    2. Jetson Platforms: TX1, TX2 and Xavier
    3. Yocto Platforms: i.MX6 and i.MX8
  6. Examples
    1. x86
      1. OpenWebRTC
      2. PubNub
      3. AppRTC
      4. MediaSoup
    2. iMX6
      1. OpenWebRTC
      2. PubNub
      3. AppRTC
    3. iMX8
      1. OpenWebRTC
      2. PubNub
      3. AppRTC
    4. TX1/TX2/Xavier
      1. OpenWebRTC
      2. PubNub
      3. AppRTC
  7. MCU Demo Application
  8. Contact Us


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