5 commits

Author	SHA1	Message	Date
Matthew Waters	50644f5718	webrtc: always reply to a promise ... Otherwise, we defeat the purpose of a promise. We were not replying when the state was closed. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>	2020-05-06 02:53:27 +00:00
Matthew Waters	cef839533e	webrtcbin: use the latest self-generated SDP as the basis for renegotiations ... Fixes multiple errors when a webrtcbin renegotiation can switch between the offerer and the answerer.	2019-07-03 23:44:15 +00:00
Matthew Waters	177aa22bcd	webrtc: Initial support for stream addition/removal ... Limitations: - No transport changes at all (ICE, DTLS) - Codec changes are untested and probably don't work - Stream removal doesn't remove transports (i.e. non-bundled transports will stay around until webrtcbin is shutdown) - Unified Plan SDP only. No Plan-B support.	2019-05-30 21:33:09 +10:00
Matthew Waters	57a006d8a5	tests/webrtc: use the existing functions in the plugin ... Instead of redefining our own, use the function implementations in webrtcsdp.c and utils.c	2018-11-26 17:13:08 +11:00
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