GstWebRTC - PubNub Audio Examples - x86: Difference between revisions

Latest revision as of 19:24, 13 April 2020

This page presents some GstRrWebRTC audio examples on x86 platform using PubNub.

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

Opus

Unidirectional elements

Example

In this example, we use rrwebrtcbin to send an audio stream and rrwebrtcbin to receive the audio stream.

Send Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler=GstPubnubSignaler \
signaler::server_url=https://webrtc.ridgerun.com:7443 \
signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink

Receive Pipeline

USER_CHANNEL=123peer
PEER_CHANNEL=123
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler=GstPubnubSignaler \
signaler::server_url=https://webrtc.ridgerun.com:7443 \
signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink

Bidirectional elements

Example

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

Send-Receive Pipeline

USER_CHANNEL=123
PEER_CHANNEL=123peer
gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler=GstPubnubSignaler \
signaler::server_url=https://webrtc.ridgerun.com:7443 \
signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
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=GstPubnubSignaler \
signaler::server_url=https://webrtc.ridgerun.com:7443 \
signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink

Previous: PubNub Signaler Examples - x86

Index

Next: PubNub Video Examples - x86

@@ Line 1: / Line 1: @@
-<table>
+{{GstWebRTC/Head|previous=PubNub Signaler Examples - x86|next=PubNub Video Examples - x86|keywords=GstRrWebRTC Examples,WebRTC Examples,GstRrWebRTC GStreamer pipelines,WebRTC GStreamer pipelines,GstRrWebRTC PubNub Signaler,WebRTC PubNub Signaler,WebRTC Audio Example, WebRTC GStreamer Audio pipelines,GstRrWebRTC PubNub,GstWebRTC Audio Example, GstRrWebRTC GStreamer Audio pipelines,WebRTC PubNub,signaling}}
-<tr>
-<td><div class="clear; float:right">__TOC__</div></td>
-<td valign=top>
-{{Debug Symbol}} Problems running the pipelines shown on this page?<br>Please see our [http://developer.ridgerun.com/wiki/index.php?title=GStreamer_Debugging GStreamer Debugging guide] for help.
-</td>
-</table>
-{{GstWebRTC Page|
+This page presents some GstRrWebRTC audio examples on '''x86 platform''' using PubNub.
-[[GstWebRTC - Audio + Video Examples|Audio + Video]]|
-[[GstWebRTC|Home]]|
-This page presents some GstWebRTC audio examples using PubNub.
+<br>
-__TOC__
+{{GStreamer debug}}
 == Opus  ==
-=== Example ===
+=== Unidirectional elements ===
-In this example we use two webrtcbins, each send an audio stream, and receives each other audio stream.
-==== x264 ====
+==== Example ====
-<pre style="background-color:yellow">
+In this example, we use rrwebrtcbin to send an audio stream and rrwebrtcbin to receive the audio stream.
-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.
-</pre>
+==== Send Pipeline ====
+<syntaxhighlight lang=bash>
+USER_CHANNEL=123
+PEER_CHANNEL=123peer
+gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=true signaler=GstPubnubSignaler \
+signaler::server_url=https://webrtc.ridgerun.com:7443 \
+signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
+audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
+</syntaxhighlight>
+==== Receive Pipeline ====
+<syntaxhighlight lang=bash>
+USER_CHANNEL=123peer
+PEER_CHANNEL=123
+gst-launch-1.0 rrwebrtcbin rtcp-mux=true start-call=false signaler=GstPubnubSignaler \
+signaler::server_url=https://webrtc.ridgerun.com:7443 \
+signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
+web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink
+</syntaxhighlight>
+=== Bidirectional elements ===
+==== Example ====
+In this example we use two rrwebrtcbins, each send an audio stream and receives each other audio stream.
 ==== Send-Receive Pipeline ====
@@ Line 31: / Line 48: @@
 USER_CHANNEL=123
 PEER_CHANNEL=123peer
-gst-launch-1.0 webrtcbin 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=GstPubnubSignaler \
+signaler::server_url=https://webrtc.ridgerun.com:7443 \
+signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
 audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink
@@ Line 41: / Line 60: @@
 USER_CHANNEL=123peer
 PEER_CHANNEL=123
-gst-launch-1.0 webrtcbin 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=GstPubnubSignaler \
+signaler::server_url=https://webrtc.ridgerun.com:7443 \
+signaler::user-channel=$USER_CHANNEL signaler::peer-channel=$PEER_CHANNEL name=web \
 audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink \
 web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink
 </syntaxhighlight>
-}}
+{{GstWebRTC/Foot|previous=PubNub Signaler Examples - x86|next=PubNub Video Examples - x86}}