It is now possible to define forward destinations for each path configuration. For each destination, the server will create a client that will forward the stream to the intended destination. Supported protocols are RTSP, RTMP, SRT. API and metrics have also been improved to allow monitoring the new forwarding system.
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
Support PROXY protocol v1/v2 on RTMP, RTMPS, RTSP, and RTSPS TCP
listeners so real client IPs are visible when running behind L4 proxies
(nginx stream, HAProxy, AWS NLB).
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
Media-over-QUIC is a streaming protocol built upon cutting edge
protocols (QUIC, HTTP3) and browser APIs (WebTransport, WebCodecs).
It's slightly faster than WebRTC, has an advanced data recovery
mechanism (placed at the frame level and not at the packet level), it
supports additional codecs (FLAC) and is less complicated to route.
Surface user agent in the auth HTTP webhook payload as "userAgent"
and in the API structs for RTMP, RTSP, WebRTC and HLS connections.
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
When MTX_QUERY is used explicitly in hooks, for instance "curl
http://something/?$MTX_QUERY", it can be used to inject arbitrary
commands. MTX_QUERY is now url-encoded to prevent any abuse regardless
of the configuration.
This fixes a long standing security flaw. Even though it's a breaking
change, few users should be impacted since this feature has been discouraged
for some time.
this allows to use MPEG-TS tracks with other protocols and with the recording system.
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
In case of RTSP and RTMP, JWTs cannot be used as passwords since
there's a size limit. Therefore, documentation is updated to suggest
passing JWT through query parameters, and authJWTInHTTPQuery is updated
to act on HTTP requests only.
Unix socket are stream-based connections, while RTP requires
packet-based connections. While packet-based Unix sockets exist
(unixgram), no client supports them. Consequently we are forced to
deprecate unix+rtp.
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.
When writing log entries to syslog, pass the level (WARN, INFO, etc)
directly to syslog. Avoid writing level and date. This provides a
better integration and allows to use syslog search tools.
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.
