Commit graph

9 commits

Author SHA1 Message Date
Ali Yousuf
69e06ced7d webrtc: Fix log when adding stun server 2019-06-04 07:54:25 +00:00
Thibault Saunier
7fe3f36ac8 Minor documentation fixes 2019-05-13 11:36:27 -04:00
Mathieu Duponchelle
85c75bb23b webrtc: expose ice-transport-policy property
This is the equivalent of iceTransportPolicy in the RTCConfiguration
dictionary.

Only two values are implemented:

* all: default behaviour
* relay: only gather relay candidates

The third member of the iceTransportPolicy enum, "public", is
obsolete.
2019-01-23 22:47:51 +00:00
Jordan Petridis
1f562870ee Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 14:18:26 +00:00
Harshad Khedkar
9ad618e487 Webrtcbin : Need to use 'host' from gst_uri_get_host s libnice agent expects it
Currently master code of gst1-plugins-bad use plain-string host name while passing it to
libnice agent: nice_agent_set_relay_info() in gstwebrtcice.c while adding turn_server(_add_turn_server).

It is observered that if we don't convert the host parameter by using gst_uri_get_host, it fails in libnice agent(0.1.14-1).

Code does, actually, set the host correctly but while passing params to nice_agent_set_relay_info, it uses incorrect one.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/823
2018-11-22 18:47:13 +05:30
Mathieu Duponchelle
45fe050286 webrtcice: do not run host resolution from applictation thread
g_resolver_lookup_by_name is a blocking call, and should not
be run when the user sets or adds a turn-server.

https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-09-19 16:17:24 +02:00
Mathieu Duponchelle
1d6160d59c webrtcbin: New add-turn-server API
It is possible and often desirable to pass multiple ICE relays
to libnice agents, the "turn-server" property, while convenient
to use from the command line, does not allow that.

This adds a new action signal, "add-turn-server" to address that.

https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-09-19 16:17:24 +02:00
Tim-Philipp Müller
6f46792f0f webrtc: Update for g_type_class_add_private() deprecation in recent GLib 2018-06-24 00:17:26 +02:00
Matthew Waters
1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00