Difference between revisions of "GstWebRTC - OpenWebRTC Web Page - x86"
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− | + | {{GstWebRTC/Head|previous=Audio + Video Examples - x86|next=Data Channel Examples - x86|keywords=GstRrWebRTC Examples,WebRTC Examples,GstRrWebRTC GStreamer pipelines,WebRTC GStreamer pipelines,OpenWebRTC signaler,OpenWebRTC Examples,OpenWebRTC web page, OpenWebRTC GStreamer pipeline,signaling}} | |
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− | {{ | + | This page presents GstRrWebRTC Web page on '''x86 platform''' to use OpenWebRTC. |
− | + | <br> | |
− | + | <br> | |
+ | {{GStreamer debug}} | ||
− | + | == 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. | |
− | + | [[File:webrtc_page.png|700px|center|Establish a WebRTC call with https://webrtc.ridgerun.com:8443]] | |
− | |||
− | [[File:webrtc_page.png|700px|center|Establish a WebRTC call with | ||
# Type a unique Session ID in the text bar. | # Type a unique Session ID in the text bar. | ||
# Select in the check box if you want audio or video streaming. | # Select in the check box if you want audio or video streaming. | ||
# Press join | # Press join | ||
− | |||
− | '''Note:''' In the following examples, the start-call property on the pipeline is set to false, | + | '''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. |
<br> | <br> | ||
<br> | <br> | ||
− | Following examples are tested using Firefox browser versions | + | 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 == | == x264 Send+Receive == | ||
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<syntaxhighlight lang=bash> | <syntaxhighlight lang=bash> | ||
− | gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstOwrSignaler signaler::server_url= | + | 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 |
+ | </syntaxhighlight> | ||
+ | |||
+ | == 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. | ||
+ | |||
+ | <syntaxhighlight lang=bash> | ||
+ | gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \ | ||
signaler::session_id=1234ridgerun name=web \ | signaler::session_id=1234ridgerun name=web \ | ||
videotestsrc is-live=true ! queue ! videoconvert ! x264enc key-int-max=1 ! rtph264pay ! queue ! web.video_sink \ | 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 | + | 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 | ||
</syntaxhighlight> | </syntaxhighlight> | ||
Line 51: | Line 57: | ||
<syntaxhighlight lang=bash> | <syntaxhighlight lang=bash> | ||
− | gst-launch-1.0 rrwebrtcbin start-call= | + | gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstOwrSignaler signaler::server_url=https://webrtc.ridgerun.com:8443 \ |
signaler::session_id=1234ridgerun name=web \ | signaler::session_id=1234ridgerun name=web \ | ||
videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \ | videotestsrc is-live=true ! vp8enc ! rtpvp8pay ! web.video_sink \ | ||
Line 57: | Line 63: | ||
</syntaxhighlight> | </syntaxhighlight> | ||
− | }} | + | == 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. | ||
+ | |||
+ | <syntaxhighlight lang=bash> | ||
+ | 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 | ||
+ | </syntaxhighlight> | ||
+ | |||
+ | </pre> | ||
+ | |||
+ | {{GstWebRTC/Foot|previous=Audio + Video Examples - x86|next=Data Channel Examples - x86}} |
Latest revision as of 14:19, 13 April 2020
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GstWebRTC | ||||||||
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WebRTC Fundamentals | ||||||||
GstWebRTC Basics | ||||||||
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Evaluating GstWebRTC | ||||||||
Getting the code | ||||||||
Building GstWebRTC | ||||||||
Examples | ||||||||
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MCU Demo Application | ||||||||
Contact Us |
This page presents GstRrWebRTC Web page on x86 platform to use OpenWebRTC.
|
Contents
SimpleRTC WebPage
The following figure show how to establish a call using the SimpleRTC web page in:
Note that browsers such as Chrome won't work with an insecure (non-https) connection.
- Type a unique Session ID in the text bar.
- Select in the check box if you want audio or video streaming.
- 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