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Nirbheek Chauhan [Thu, 8 Mar 2018 14:40:55 +0000 (20:10 +0530)]
Fix crash on Windows by delimiting option entries with NULL
Also use more verbose forms of g_assert which print values on failure
Nirbheek Chauhan [Sat, 17 Feb 2018 02:40:59 +0000 (08:10 +0530)]
README: link to blog post, document multiparty example
Also add TODO stubs for MCU and SFU
Tim-Philipp Müller [Fri, 2 Feb 2018 08:41:21 +0000 (08:41 +0000)]
README: fix formatting
Tim-Philipp Müller [Fri, 2 Feb 2018 08:39:04 +0000 (08:39 +0000)]
webrtc-sendrecv: define GST_USE_UNSTABLE_API to avoid compiler warnings
Tim-Philipp Müller [Fri, 2 Feb 2018 08:23:30 +0000 (08:23 +0000)]
Update README
Point to upstream repos now that it's been merged
Nirbheek Chauhan [Tue, 12 Dec 2017 16:10:09 +0000 (21:40 +0530)]
sendrecv: Add a Google STUN server to the configuration
Without this, the example will only work on link-local and localhost
networks.
Matthew Waters [Wed, 22 Nov 2017 13:21:36 +0000 (00:21 +1100)]
server/js: also allow running on localhost
Mathieu Duponchelle [Wed, 22 Nov 2017 12:15:48 +0000 (13:15 +0100)]
Update to new promise API
Nirbheek Chauhan [Mon, 30 Oct 2017 07:54:21 +0000 (13:24 +0530)]
multiparty sendrecv: Add a queue before the audio sink
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.
Nirbheek Chauhan [Mon, 30 Oct 2017 03:39:36 +0000 (09:09 +0530)]
WIP: Add a new multiparty sendrecv gstreamer demo
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
Nirbheek Chauhan [Mon, 30 Oct 2017 03:42:06 +0000 (09:12 +0530)]
sendrecv: Rename function for greater clarity
Nirbheek Chauhan [Sat, 28 Oct 2017 22:38:45 +0000 (04:08 +0530)]
Update Protocol.md
Fix indentation typos
Nirbheek Chauhan [Sat, 28 Oct 2017 13:30:42 +0000 (19:00 +0530)]
simple-server: Add support for multi-party rooms
Also add a new room-client.py to test the protocol which is documented
in Protocol.md
Nirbheek Chauhan [Sat, 28 Oct 2017 13:32:56 +0000 (19:02 +0530)]
Protocol.md: Fix headings
Nirbheek Chauhan [Sat, 28 Oct 2017 13:30:03 +0000 (19:00 +0530)]
signalling/client.py: Rename to session-client.py
Also fix CALL -> SESSION naming
Nirbheek Chauhan [Sat, 21 Oct 2017 14:27:29 +0000 (19:57 +0530)]
Add sendrecv implementation in js and gst webrtc
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/
Nirbheek Chauhan [Sat, 21 Oct 2017 14:26:52 +0000 (19:56 +0530)]
Add a simple python3 webrtc signalling server
+ client for testing + protocol documentation
Nirbheek Chauhan [Sat, 21 Oct 2017 14:13:01 +0000 (19:43 +0530)]
Initial commit