Joshua Piccari e4fe25fdb6 webrtc: add liveness check to WebRTC output using data channel ...

Previously, there was no way to detect if WebRTC clients were still
connected to the stream. This lead to the RTC streams being kept
open indefinitely when clients focibly closed the stream. This can
happen with Chrome, Edge, Safari, etc when the tab is closed or on
mobile devices when the screen is locked or browser closed.

This change adds a data channel to the stream and requires users to
respond to ping requests. While sending a frame the ping might be
sent to the client. If clients do not respond to these ping requests before
the timeout duration the stream is closed and the client is removed
from the server. This causes the client to reinitiate the connection.

The timeout defaults to 30 seconds.

The client needs to send `keepAlive: true` as part of webrtc request.
On the received `ping` send back the `pong` over data-channel.

2025-07-03 00:07:59 +02:00

..

rtsp

rtsp: disable RTSP support if using old livemedia version

2025-07-02 20:55:47 +02:00

webrtc

webrtc: add liveness check to WebRTC output using data channel

2025-07-03 00:07:59 +02:00

http_ffmpeg.c

Add -snapshot.height, -stream.height and -video.height

2023-02-22 17:32:46 +01:00

http_h264.c

Add -snapshot.height, -stream.height and -video.height

2023-02-22 17:32:46 +01:00

http_hls.c

Improve HTTP support (to better handle params) and redirect /video params

2022-09-24 10:41:09 +02:00

http_jpeg.c

http: fix potential memory corruption

2023-06-02 00:14:41 +02:00

output.c

Ensure that /snapshot is always "up-to date"

2023-04-29 10:16:04 +02:00

output.h

Add -snapshot.height, -stream.height and -video.height

2023-02-22 17:32:46 +01:00