Commit graph

37 commits

Author SHA1 Message Date
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
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
Based on https://github.com/ttustonic/GStreamerSharpSamples from
Tomislav Tustonić <ttustonic@outlook.com>
2018-07-11 10:05:38 +00:00
Leon Tan
b6300d3b92 Fix bug in Rust sendrecv demo 2018-06-27 22:58:19 +02:00
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
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
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
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
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
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
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
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
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
Tim-Philipp Müller
72c10e8243 webrtc-sendrecv: define GST_USE_UNSTABLE_API to avoid compiler warnings 2018-02-02 08:39:04 +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
569aff43f9 sendrecv: Rename function for greater clarity 2017-10-30 09:14:29 +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