GstWebRTC - OpenWebRTC Web Page - iMX6

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 some GstRrWebRTC Web page on IMX6 platform to use OpenWebRTC.

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

Connect to SimpleRTC WebPage

To run the following examples, first you need to open the SimpleRTC web page in the browser.

• https://webrtc.ridgerun.com:8443

The following figure show how to establish a call using the SimpleRTC web page in 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 to start the call after you joined in the website.

Following examples are tested using Firefox browser version 64.0 (64-bit) and Chrome browser version 71.0.3 (64-bit) for testing the demo OpenWebRTC web page.

OPUS Send

Example

This pipeline will send an audio stream to the demo web page.

gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \
signaler::session_id=1234ridgerun name=web \
audiotestsrc is-live=true wave=8 ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! web.audio

H264 Send

Example

This pipeline will encode a video stream to H264 and send it to the demo web page.

We set the profile-level-id because the imxvpuenc_h264 don't set the profile and level for that reason the rtph264pay don't set the profile-level-id

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 ! imxvpuenc_h264 ! rtph264pay ! \
capssetter caps="application/x-rtp,profile-level-id=(string)42c01f" ! web.video_sink

H264+OPUS Send

Example

This pipeline will send a video stream and an audio stream to the demo web page.

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 ! imxvpuenc_h264 ! rtph264pay ! \
capssetter caps="application/x-rtp,profile-level-id=(string)42c01f" ! web.video_sink
audiotestsrc is-live=true wave=8 ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! web.audio

H264 Send+Receive

We set the profile-level-id because the imxvpuenc_h264 don't set the profile and level for that reason the rtph264pay don't set the profile-level-id

Example

This pipeline will send a video stream and an audio stream to the demo web page. Additionally, it will receive the web page's video and audio feeds, 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=testingrr name=web \
videotestsrc is-live=true ! imxvpuenc_h264 ! rtph264pay ! \ 
capssetter caps="application/x-rtp,profile-level-id=(string)42c01f" ! web.video_sink web.video_src ! rtph264depay ! h264parse ! imxvpudec ! queue ! imxg2dvideosink sync=false

IMX6 Limitations

GstRrWebRTC was originally developed for x86 platform, but it's supported for IMX6 with some limitations:

• Only true start-call supported: The pipeline needs to start the call to allow WebRTC streaming.
• Required Capssetter: For IMX6 implementations it's necessary to select an specific profile level id in order to execute the video streaming.