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