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Author SHA1 Message Date
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
Fix indentation typos
 2017-10-29 04:08:45 +05:30
Nirbheek Chauhan
d687ff3d91 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
 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 
Also fix CALL -> SESSION naming
 2017-10-28 19:00:03 +05:30
Nirbheek Chauhan
e9b0656bad 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/
 2017-10-21 20:02:19 +05:30
Nirbheek Chauhan
663ad7ba98 Add a simple python3 webrtc signalling server 
+ 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