Commit graph

4 commits

Author SHA1 Message Date
Olivier Crête
df8d29e9c3 webrtcbin: Remove remnant of non-rtcp-mux mode
There was some code left that wasn't used anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1930>
2021-01-06 23:02:37 +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
Jan Schmidt
15d3bc9870 webrtc: Clean up and fix transportsendbin
Refactor transportsendbin, and change the way
pads are blocked on dtlssrtpenc so that they
don't interfere with state changes.

As well as being easier to read, this fixes
spurious failures shutting down webrtcbin
if DTLS negotiation hasn't completed yet.
2018-07-14 23:20:13 +10:00
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