mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-05-10 20:32:43 +00:00
83d70d3471
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization) for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to identify the clock. These attributes are handled by `rtpjitterbuffer` on the consumer side. They MUST be part of the SDP offer. When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the clock's origin. Because the payloaders are not instantiated when the offer is sent to the consumer, the RTP offset is set to 0 and the payloader `timstamp-offset`s are set accordingly when they are created. The `webrtc-precise-sync` examples were updated to be able to start with an NTP (default), a PTP or the system clock (on the receiver only). The rtp jitter buffer will synchronize with the clock signalled in the SDP offer provided the sender is started with `--do-clock-signalling` & the receiver with `--expect-clock-signalling`. [RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500> |
||
|---|---|---|
| .. | ||
| aws | ||
| hlssink3 | ||
| ndi | ||
| onvif | ||
| raptorq | ||
| reqwest | ||
| rtp | ||
| rtsp | ||
| webrtc | ||
| webrtchttp | ||