Commit graph

6 commits

Author SHA1 Message Date
Olivier Crête
52c676546d webrtc: Also remove rtcp_transport from the structure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
2020-11-24 01:59:55 +00:00
Olivier Crête
c5d76d944e webrtc: Remove APIs to set transport on sender/receiver
They're not not used ever.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
2020-11-24 01:59:55 +00:00
Olivier Crête
5d5417f271 webrtc: Remove non rtcp-mux code
RTCP mux is now always required by the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
2020-11-24 01:59:55 +00:00
Mathieu Duponchelle
42adb02a10 docstrings: port ulinks to markdown links 2019-08-23 20:14:12 +02:00
Sebastian Dröge
950ead9215 webrtc: Add some locks to setters and remove non-existing functions from headers
https://bugzilla.gnome.org/show_bug.cgi?id=794363
2018-03-16 10:37:24 +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