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
