When the publisher or source of a stream is offline, the server can be
configured to fill gaps in the stream with a video that is played on
repeat until a publisher comes back online. This allows readers to stay
connected regardless of the state of the stream. The offline video and
any future online stream are concatenated without decoding or
re-encoding packets, using the original codec.
this allows to apply features that were previously implemented for
single codecs (like RTP packet resizing), to any codec, and simplifies
future development.
Absolute timestamps are used to generate recording segments, so users
should not be able to change them, unless explicitly allowed.
Parameter useAbsoluteTimestamp allowed to re-enable absolute timestamps
with RTSP and WebRTC, now it is extended to all protocols.
When the absolute timestamp of incoming frames was not available, it
was filled with the current timestamp, which is influenced by latency
over time.
This mechanism is replaced by an algorithm that detects when latency is
the lowest, stores the current timestamp and uses it as reference
throughout the rest of the stream.
in API (/paths/list, /paths/get) and metrics (paths_bytes_sent), the
amount of sent bytes was increased even in case of writes to the
recorder and HLS muxer, which are not generating network traffic. This
fixes the issue.
Discarded frames, decode errors and lost packets were logged
individually, then there was a mechanism that prevented more than 1 log
entry per second from being printed, resulting in inaccurate reports.
Now discarded frames, decode errors and lost packets are accurately
counted, and their count is printed once every second.
