Nirbheek Chauhan
3879a5078d
sendrecv/js: Explicitly close the local stream when done
...
This immediately releases the webcam and mic instead of lazily at some
unpredictable time in the future.
2018-04-01 01:10:00 +05:30
Nirbheek Chauhan
3eabe5cb0b
sendrecv/js: Make error statuses more prominent
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Colour errors in red, and ensure that later status updates don't
overwrite existing error statuses.
2018-04-01 01:09:54 +05:30
Nirbheek Chauhan
bd6deaca46
sendrecv/js: Call getUserMedia on incoming call
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Instead of registering it on page load. This will allow us to add an
option for users to override the default constraints later.
This is also generally nicer because the browser won't open the webcam
immediately when you load the page and keep recording from it.
2018-04-01 01:09:46 +05:30
Nirbheek Chauhan
563826deaf
sendrecv: Don't set pipeline state if it's NULL
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Avoids ugly CRITICAL warnings when erroring out.
2018-03-31 10:28:51 +05:30
Nirbheek Chauhan
82314cabbb
Don't use strict ssl certificate checking for localhost
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When using localhost signalling servers, we don't want to use
strict ssl because it's probably using a self-signed certificate
and there's no need to do certificate checking over localhost anyway.
2018-03-31 10:27:05 +05:30
Nirbheek Chauhan
0e1be2a63f
Add Makefiles for all C demos
2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
2d2bc0fe0e
Fix compiler warnings in all C demos
2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
20cf2503ee
sendrecv: Fix SDP message format
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The format is {'sdp': {'sdp': <sdp>, 'type': <sdptype>}}
The multiparty-sendrecv demo already uses this format.
2018-03-23 19:00:37 +05:30
Sebastian Kilb
2b82525bb0
Fix audio/video linking error on windows
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Closes https://github.com/centricular/gstwebrtc-demos/issues/5
2018-03-21 06:26:49 +05:30
Nirbheek Chauhan
429f4bb65f
README.md: Document the binaries and Cerbero
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Also mention where to file bug reports about the plugin itself.
2018-03-10 13:21:34 +05:30
Nirbheek Chauhan
55e86469d9
Check for all necessary plugins at startup
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People seem to be having problems ensuring that they have all the
right plugins built, so make it a bit easier for them.
2018-03-10 01:54:48 +05:30
Nirbheek Chauhan
fa2adc717b
Fix crash on Windows by delimiting option entries with NULL
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Also use more verbose forms of g_assert which print values on failure
2018-03-08 20:10:55 +05:30
Nirbheek Chauhan
492d13a7c9
README: link to blog post, document multiparty example
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Also add TODO stubs for MCU and SFU
2018-02-17 08:13:36 +05:30
Tim-Philipp Müller
2e5204ae3b
README: fix formatting
2018-02-02 08:41:21 +00:00
Tim-Philipp Müller
72c10e8243
webrtc-sendrecv: define GST_USE_UNSTABLE_API to avoid compiler warnings
2018-02-02 08:39:04 +00:00
Tim-Philipp Müller
43a27385c3
Update README
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Point to upstream repos now that it's been merged
2018-02-02 08:23:30 +00:00
Nirbheek Chauhan
97cf763420
sendrecv: Add a Google STUN server to the configuration
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Without this, the example will only work on link-local and localhost
networks.
2017-12-12 21:40:48 +05:30
Matthew Waters
e4e83a648b
server/js: also allow running on localhost
2017-11-23 00:29:39 +11:00
Mathieu Duponchelle
e5c5767298
Update to new promise API
2017-11-22 22:28:55 +10:00
Nirbheek Chauhan
0c5e799952
multiparty sendrecv: Add a queue before the audio sink
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Missed this, fixes the bug where removing a peer causes the pipeline to
get stuck. However, when peers leave, there is still a chance that the
pipeline will get stuck.
2017-10-30 13:24:21 +05:30
Nirbheek Chauhan
9b1a0e5389
WIP: Add a new multiparty sendrecv gstreamer demo
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You can join a room and an audio-only call will be started with all
peers in that room. Currently uses audiotestsrc only.
BUG: With >2 peers in a call, if a peer leaves, the pipeline stops
outputting data from the remaining peers to the (audio) sink.
TODO: JS code to allow a browser to join the call
TODO: Cleanup pipeline when a peer leaves
TODO: Add ICE servers to allow calls over the Internet
TODO: Perhaps setup a TURN server as well
2017-10-30 09:14:29 +05:30
Nirbheek Chauhan
569aff43f9
sendrecv: Rename function for greater clarity
2017-10-30 09:14:29 +05:30
Nirbheek Chauhan
96e4f39fd8
Update Protocol.md
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Fix indentation typos
2017-10-29 04:08:45 +05:30
Nirbheek Chauhan
d687ff3d91
simple-server: Add support for multi-party rooms
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Also add a new room-client.py to test the protocol which is documented
in Protocol.md
2017-10-28 19:20:44 +05:30
Nirbheek Chauhan
2db85c41cc
Protocol.md: Fix headings
2017-10-28 19:03:11 +05:30
Nirbheek Chauhan
c2961305e3
signalling/client.py: Rename to session-client.py
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Also fix CALL -> SESSION naming
2017-10-28 19:00:03 +05:30
Nirbheek Chauhan
e9b0656bad
Add sendrecv implementation in js and gst webrtc
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JS code runs on the browser and uses the browser's webrtc
implementation.
C code uses gstreamer's webrtc implementation, for which you need the
following repositories:
https://github.com/ystreet/gstreamer/tree/promise
https://github.com/ystreet/gst-plugins-bad/tree/webrtc
You can build these with either Autotools gst-uninstalled:
https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/
Or with Meson gst-build:
https://cgit.freedesktop.org/gstreamer/gst-build/
2017-10-21 20:02:19 +05:30
Nirbheek Chauhan
663ad7ba98
Add a simple python3 webrtc signalling server
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+ client for testing + protocol documentation
2017-10-21 19:56:52 +05:30
Nirbheek Chauhan
8d782e4460
Initial commit
2017-10-21 19:43:01 +05:30
