Jan Alexander Steffens (heftig)
fd1d53b04a
on_server_message: Do not unref message GBytes
...
We don't own the reference. Since GLib 2.58, the g_bytes_unref that
follows the signal emission in libsoup loudly complains about the
attempt to underflow the refcount.
2018-09-21 13:12:43 +00:00
Mathieu Duponchelle
547f296293
sendrecv: try to add a data channel
2018-09-21 13:12:16 +00:00
Mathieu Duponchelle
7865c31387
webrtc.js: fix tearing down
2018-09-21 13:12:16 +00:00
Sebastian Dröge
fe6267fe0d
Update to releases of glib/gstreamer bindings
2018-09-10 14:06:01 +03:00
meldron
dc1163ab95
Fix stun server address
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The stun server address has a space as suffix which is not allowed in the rust bindings.
2018-07-26 12:11:37 +00:00
Thibault Saunier
122c4106a4
Implement the demo in C# with GStreamerSharp
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Based on https://github.com/ttustonic/GStreamerSharpSamples from
Tomislav Tustonić <ttustonic@outlook.com>
2018-07-11 10:05:38 +00:00
Nirbheek Chauhan
c5e5a7cfd3
Update README.md
2018-07-03 19:26:56 +05:30
Leon Tan
b6300d3b92
Fix bug in Rust sendrecv demo
2018-06-27 22:58:19 +02:00
Matthew Clark
37e8853041
Correct signalling usage instructions
2018-06-27 01:29:54 +00:00
Mathieu Duponchelle
1958814680
webrtc-sendrecv.py: required gstreamer 1.14.2
...
Addresses #25
2018-06-25 14:45:57 +02:00
Sebastian Dröge
9cf3aa088e
General code cleanup of the Rust sendrecv demo
...
Fewer clones and more borrowing, if let instead of match, match instead
of multiple ifs, insert a few newlines all over the place to make code
less dense, and a few changes to make code a bit more idiomatic.
2018-06-21 13:16:15 +03:00
Sebastian Dröge
2614249149
Fix various clippy warnings in the Rust sendrecv demo
2018-06-21 09:03:18 +03:00
maxmcd
b826f968cb
Add --disable-ssl flag to webrtc-sendrecv.c
2018-06-18 09:02:05 +03:00
maxmcd
83b9c4efd7
Add --disable-ssl option to simple-server.py
2018-06-18 09:02:05 +03:00
maxmcd
bb56d6eab7
Add Rust version of sendrecv example
...
This also comes with a docker image to collect all dependencies and
build everything.
Fixes https://github.com/centricular/gstwebrtc-demos/pull/20
2018-06-18 09:02:05 +03:00
Mathieu Duponchelle
3603899291
webrtc-sendrecv.py: improve debug and documentation
2018-06-11 20:26:07 +02:00
Mathieu Duponchelle
56c17d6487
sendrecv: python version
2018-06-11 18:49:53 +02:00
Nirbheek Chauhan
bba6c92392
Fix heading levels
2018-04-11 19:04:47 +05:30
Eloi Bail
d6741c1f80
mp-webrtc-sendrecv.c: add missing comma in the list of package required
...
A comma is missing in the list of package required. Thus the package
'srtprtpmanager' is checked instead of packages srtp and rtpmanager.
2018-04-03 15:04:57 +00:00
Nirbheek Chauhan
ea8e960e29
sendrecv/js: Improve more logging and errors
2018-04-01 01:53:44 +05:30
Nirbheek Chauhan
9cc57d2dd1
sendrecv/js: Fix some null/undefined checks
2018-04-01 01:52:46 +05:30
Nirbheek Chauhan
669d234ebd
sendrecv/js: Don't reuse peer_id across sessions
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It increases the likelihood of a collision with someone else, and it
was an unintended side-effect anyway.
2018-04-01 01:30:23 +05:30
Nirbheek Chauhan
47bfa3cc27
sendrecv/gst: Add no-op audio/video converters
...
This reduces the chance that someone will try to change the
audio/video source elements and get an error because they don't know
about the conversion elements. They will be no-ops in the usual case.
Closes https://github.com/centricular/gstwebrtc-demos/issues/8
2018-04-01 01:15:16 +05:30
Nirbheek Chauhan
7c5fbf1aca
sendrecv/js: custom getUserMedia constraints
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The html page now contains a text area in which the default
constraints will be added and can be edited.
Closes https://github.com/centricular/gstwebrtc-demos/issues/11
2018-04-01 01:10:22 +05:30
Nirbheek Chauhan
fe40c70536
sendrecv/js: Simplify local stream management
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Just use the fulfilled value of the promise directly instead of
storing it separately
2018-04-01 01:10:09 +05:30
Nirbheek Chauhan
9f4783fb60
sendrecv/js: Allow overriding peer_id and ws_server
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This allows people to easily use a custom peer id or their own server
if the automatic values are not appropriate for them.
2018-04-01 01:10:09 +05:30
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
...
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
...
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
...
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
...
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