GstWebRTC - OpenWebRTC Web Page - x86

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

This page presents GstRrWebRTC Web page on x86 platform to use OpenWebRTC.

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

SimpleRTC WebPage

The following figure show how to establish a call using the SimpleRTC web page in:

• https://webrtc.ridgerun.com:8443 (recommended)
• http://webrtc.ridgerun.com:8080

Note that browsers such as Chrome won't work with an insecure (non-https) connection.

Establish a WebRTC call with https://webrtc.ridgerun.com:8443

1. Type a unique Session ID in the text bar.
2. Select in the check box if you want audio or video streaming.
3. Press join

Note: In the following examples, the start-call property on the pipeline is set to true, thus the pipeline starts the call. If start-call is set to false, you have to start the call from the website pressing the correct button.

Following examples are tested using Firefox browser versions 64.0 and 71.0.3 (64-bit) for testing the demo OpenWebRTC web page. These pipelines will not work with the newer versions of Firefox browser and also it will not work with the Chrome browser.

x264 Send+Receive

Example

This pipeline will encode a video stream to H264 and send it to the demo web page. Additionally, it will receive the web page's video feed, in the same format.

gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 signaler::session_id=1234ridgerun name=web videotestsrc is-live=true ! queue ! videoconvert ! x264enc key-int-max=1 ! rtph264pay ! queue ! web.video_sink web.video_src ! rtph264depay ! avdec_h264 ! videoconvert ! ximagesink async=true sync=false enable-last-sample=false

x264+OPUS Send+Receive

Example

This pipeline will send an audio and a H264 video streams the demo web page. Additionally, it will receive the web page's streams, in the same format.

gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \
signaler::session_id=1234ridgerun name=web \
videotestsrc is-live=true ! queue ! videoconvert ! x264enc key-int-max=1 ! rtph264pay ! queue ! web.video_sink \
web.video_src ! rtph264depay ! avdec_h264 ! videoconvert ! ximagesink async=true \
audiotestsrc is-live=true wave=8 ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! web.audio_sink \
web.audio_src ! rtpopusdepay ! opusdec ! audioconvert ! alsasink async=false

VP8 Send+Receive

Example

This pipeline will encode a video stream to VP8 and send it to the demo web page. Additionally, it will receive the web page's video feed, in the same format.

gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \
signaler::session_id=1234ridgerun name=web \
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \
web.video_src ! rtpvp8depay ! vp8dec ! videoconvert ! ximagesink async=true

VP8+OPUS Send+Receive

Example

This pipeline will send an audio and a Vp8 video streams the demo web page. Additionally, it will receive the web page's streams, in the same format.

gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \
signaler::session_id=1234ridgerun name=web \
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \
web.video_src ! rtpvp8depay ! vp8dec ! videoconvert ! ximagesink async=true \
audiotestsrc is-live=true wave=8 ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! web.audio_sink \
web.audio_src ! rtpopusdepay ! opusdec ! audioconvert ! alsasink async=false