Commit graph

8 commits

Author SHA1 Message Date
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
8e3472faee webrtc: Use the dtlssrtenc rtp-sync property
Instead of synchronising at the ICE transport, do clock sync for the
RTP stream at the DTLS transport via the dtlssrtpenc rtp-sync
property. This avoids delaying RTCP while waiting until it is time
to output an RTP packet when rtcp-mux is enabled.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
2020-02-27 12:30:32 +00:00
Sebastian Dröge
b2e7739364 webrtc/dtlstransport: Proxy DTLS connection state from the DTLS elements to the transport
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Mathieu Duponchelle
42adb02a10 docstrings: port ulinks to markdown links 2019-08-23 20:14:12 +02:00
Jan Schmidt
cb750efd6c webrtc: Move dtlssrtpenc state management
Move the errant piece of dtlssrtpenc state change
management from dtlstransport in the Webrtc libs,
into the transportsendbin that does the rest of
the element management so it's all in one place.
2018-07-14 23:18:50 +10:00
Jan Schmidt
3a6777d599 webrtc/dtlstransport: Add more debug. Rename category
Rename the dtlstransport debug category to webrtcdtlstransport.
2018-07-14 23:18:40 +10:00
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