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support publishing, reading, proxying with SRT (#2068)
This commit is contained in:
Alessandro Ros
GitHub
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d696a782f7
commit
b4e3033ea3
22 changed files with 2184 additions and 213 deletions
119
README.md
119
README.md
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@ -20,9 +20,11 @@ Live streams can be published to the server with:
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|protocol|variants|video codecs|audio codecs|
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|--------|--------|------------|------------|
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|[WebRTC](#webrtc)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711|
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|[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
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|[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
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|[SRT clients](#srt-clients)||H265, H264|Opus, MPEG-4 Audio (AAC)|
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|[SRT servers](#srt-servers)||H265, H264|Opus, MPEG-4 Audio (AAC)|
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|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711|
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|[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec|
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|[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec|
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|[RTMP clients](#rtmp-clients)|RTMP, RTMPS, Enhanced RTMP|AV1, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
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|[RTMP cameras and servers](#rtmp-cameras-and-servers)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
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|[HLS cameras and servers](#hls-cameras-and-servers)|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
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@ -33,8 +35,9 @@ And can be read from the server with:
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|protocol|variants|video codecs|audio codecs|
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|--------|--------|------------|------------|
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|[WebRTC](#webrtc-1)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711|
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|[RTSP](#rtsp)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
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|[SRT](#srt)||H265, H264|Opus, MPEG-4 Audio (AAC)|
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|[WebRTC](#webrtc)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711|
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|[RTSP](#rtsp)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec|
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|[RTMP](#rtmp)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
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|[HLS](#hls)|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
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@ -76,7 +79,9 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
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* [Generic webcam](#generic-webcam)
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* [Raspberry Pi Cameras](#raspberry-pi-cameras)
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* [By protocol](#by-protocol)
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* [WebRTC](#webrtc)
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* [SRT clients](#srt-clients)
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* [SRT servers](#srt-servers)
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* [WebRTC clients](#webrtc-clients)
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* [RTSP clients](#rtsp-clients)
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* [RTSP cameras and servers](#rtsp-cameras-and-servers)
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* [RTMP clients](#rtmp-clients)
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@ -90,7 +95,8 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
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* [VLC](#vlc)
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* [Web browsers](#web-browsers-1)
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* [By protocol](#by-protocol-1)
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* [WebRTC](#webrtc-1)
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* [SRT](#srt)
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* [WebRTC](#webrtc)
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* [RTSP](#rtsp)
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* [RTMP](#rtmp)
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* [HLS](#hls)
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@ -157,6 +163,7 @@ docker run --rm -it \
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-p 1935:1935 \
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-p 8888:8888 \
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-p 8889:8889 \
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-p 8890:8890/udp \
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bluenviron/mediamtx
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```
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@ -243,7 +250,7 @@ makepkg -si
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#### FFmpeg
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FFmpeg can publish a stream to the server in multiple ways (RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
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FFmpeg can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
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```
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ffmpeg -re -stream_loop -1 -i file.ts -c copy -f rtsp rtsp://localhost:8554/mystream
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@ -259,7 +266,7 @@ The resulting stream will be available in path `/mystream`.
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#### GStreamer
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GStreamer can publish a stream to the server in multiple ways (RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
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GStreamer can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
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```sh
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gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream \
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@ -286,7 +293,7 @@ The resulting stream will be available in path `/mystream`.
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#### OBS Studio
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OBS Studio can publish to the server as a [RTMP client](#rtmp-clients). In `Settings -> Stream` (or in the Auto-configuration Wizard), use the following parameters:
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OBS Studio can publish to the server in multiple ways (SRT client, RTMP client, WebRTC client). The recommended one consists in publishing as a [RTMP client](#rtmp-clients). In `Settings -> Stream` (or in the Auto-configuration Wizard), use the following parameters:
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* Service: `Custom...`
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* Server: `rtmp://localhost`
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@ -324,7 +331,7 @@ The resulting stream will be available in path `/mystream`.
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#### OpenCV
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OpenCV can publish to the server as a [RTSP client](#rtsp-clients). It must be compiled with GStreamer support, by following this procedure:
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OpenCV can publish to the server through its GStreamer plugin, as a [RTSP client](#rtsp-clients). It must be compiled with GStreamer support, by following this procedure:
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```sh
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sudo apt install -y libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev gstreamer1.0-plugins-ugly gstreamer1.0-rtsp python3-dev python3-numpy
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@ -535,7 +542,38 @@ The resulting stream will be available in path `/cam_with_audio`.
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### By protocol
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#### WebRTC
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#### SRT clients
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SRT is a protocol that allows to publish and read live data stream, providing encryption, integrity and a retransmission mechanism. It is usually used to transfer media streams encoded with MPEG-TS. In order to publish a stream to the server with the SRT protocol, use this URL:
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```
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srt://localhost:8890?streamid=publish:mystream&pkt_size=1316
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```
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Replace `mystream` with any name you want. The resulting stream will be available in path `/mystream`.
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If credentials are enabled, append username and password to `streamid`;
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```
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srt://localhost:8890?streamid=publish:mystream:user:pass&pkt_size=1316
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```
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If you want to publish a stream by using a client in listening mode (i.e. with `mode=listener` appended to the URL), read the next section.
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Known clients that can publish with SRT are [FFmpeg](#ffmpeg), [Gstreamer](#gstreamer), [OBS Studio](#obs-studio).
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#### SRT servers
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In order to ingest into the server a SRT stream from an existing server, camera or client in listening mode (i.e. with `mode=listener` appended to the URL), add the corresponding URL into the `source` parameter of a path:
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```yml
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paths:
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proxied:
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# url of the source stream, in the format srt://host:port?streamid=streamid&other_parameters
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source: srt://original-url
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```
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#### WebRTC clients
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WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can publish a stream with WebRTC and a web browser by visiting:
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@ -553,9 +591,11 @@ http://localhost:8889/mystream/whip
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Depending on the network it may be difficult to establish a connection between server and clients, see [WebRTC-specific features](#webrtc-specific-features) for remediations.
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Known clients that can publish with WebRTC and WHIP are [FFmpeg](#ffmpeg), [Gstreamer](#gstreamer), [OBS Studio](#obs-studio).
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#### RTSP clients
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RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see [RTSP-specific features](#rtsp-specific-features)). In order to publish a stream with the RTSP protocol, you can use this URL:
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RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see [RTSP-specific features](#rtsp-specific-features)). In order to publish a stream to the server with the RTSP protocol, use this URL:
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```
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rtsp://localhost:8554/mystream
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@ -563,6 +603,8 @@ rtsp://localhost:8554/mystream
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The resulting stream will be available in path `/mystream`.
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Known clients that can publish with RTSP are [FFmpeg](#ffmpeg), [Gstreamer](#gstreamer), [OBS Studio](#obs-studio).
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#### RTSP cameras and servers
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Most IP cameras expose their video stream by using a RTSP server that is embedded into the camera itself. You can use _MediaMTX_ to connect to one or multiple existing RTSP servers and read their video streams:
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@ -603,6 +645,8 @@ In case authentication is enabled, credentials can be passed to the server by us
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rtmp://localhost/mystream?user=myuser&pass=mypass
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```
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Known clients that can publish with RTMP are [FFmpeg](#ffmpeg), [Gstreamer](#gstreamer), [OBS Studio](#obs-studio).
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#### RTMP cameras and servers
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You can use _MediaMTX_ to connect to one or multiple existing RTMP servers and read their video streams:
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@ -657,13 +701,15 @@ paths:
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The resulting stream will be available in path `/mypath`.
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Known clients that can publish with WebRTC and WHIP are [FFmpeg](#ffmpeg) and [Gstreamer](#gstreamer).
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## Read from the server
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### By software
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#### FFmpeg
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FFmpeg can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP). The recommended one consists in reading with [RTSP](#rtsp):
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FFmpeg can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP, SRT). The recommended one consists in reading with [RTSP](#rtsp):
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```sh
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ffmpeg -i rtsp://localhost:8554/mystream -c copy output.mp4
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@ -677,7 +723,7 @@ ffmpeg -rtsp_transport tcp -i rtsp://localhost:8554/mystream -c copy output.mp4
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#### GStreamer
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GStreamer can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP). The recommended one consists in reading with [RTSP](#rtsp):
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GStreamer can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP, SRT). The recommended one consists in reading with [RTSP](#rtsp):
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```sh
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gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/mystream latency=0 ! decodebin ! autovideosink
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@ -697,7 +743,7 @@ gst-launch-1.0 rtspsrc tls-validation-flags=0 location=rtsps://ip:8322/...
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#### VLC
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VLC can read a stream from the server in multiple ways (RTSP, RTMP, HLS). The recommended one consists in reading with [RTSP](#rtsp):
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VLC can read a stream from the server in multiple ways (RTSP, RTMP, HLS, SRT). The recommended one consists in reading with [RTSP](#rtsp):
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```sh
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vlc --network-caching=50 rtsp://localhost:8554/mystream
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@ -764,6 +810,24 @@ This web page can be embedded into another web page by using an iframe:
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### By protocol
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#### SRT
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SRT is a protocol that allows to publish and read live data stream, providing encryption, integrity and a retransmission mechanism. It is usually used to transfer media streams encoded with MPEG-TS. In order to read a stream from the server with the SRT protocol, use this URL:
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```
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srt://localhost:8890?streamid=read:mystream&pkt_size=1316
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```
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Replace `mystream` with the path name.
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If credentials are enabled, append username and password to `streamid`;
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```
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srt://localhost:8890?streamid=publish:mystream:user:pass&pkt_size=1316
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```
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Known clients that can read with SRT are [FFmpeg](#ffmpeg-1), [Gstreamer](#gstreamer-1) and [VLC](#vlc).
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#### WebRTC
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WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can read a stream with WebRTC and a web browser by visiting:
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@ -780,14 +844,18 @@ http://localhost:8889/mystream/whep
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Depending on the network it may be difficult to establish a connection between server and clients, see [WebRTC-specific features](#webrtc-specific-features) for remediations.
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Known clients that can read with WebRTC and WHEP are [FFmpeg](#ffmpeg-1), [Gstreamer](#gstreamer-1) and [web browsers](#web-browsers-1).
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#### RTSP
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RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see [RTSP-specific features](#rtsp-specific-features)). In order to read a stream with the RTSP protocol, you can use this URL:
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RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see [RTSP-specific features](#rtsp-specific-features)). In order to read a stream with the RTSP protocol, use this URL:
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```
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rtsp://localhost:8554/mystream
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```
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Known clients that can read with SRT are [FFmpeg](#ffmpeg-1), [Gstreamer](#gstreamer-1) and [VLC](#vlc).
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##### Latency
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The RTSP protocol doesn't introduce any latency by itself. Latency is usually introduced by clients, that put frames in a buffer to compensate network fluctuations. In order to decrease latency, the best way consists in tuning the client. For instance, latency can be decreased with VLC by decreasing the Network caching parameter, that is available in the Open network stream dialog or alternatively ca be set with the command line:
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rtmp://localhost/mystream?user=myuser&pass=mypass
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```
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Known clients that can read with RTMP are [FFmpeg](#ffmpeg-1), [Gstreamer](#gstreamer-1) and [VLC](#vlc).
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#### HLS
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HLS is a protocol that works by splitting streams into segments, and by serving these segments and a playlist with the HTTP protocol. You can use _MediaMTX_ to generate a HLS stream, that is accessible through a web page:
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@ -824,9 +894,9 @@ and can also be accessed without using the browsers, by software that supports t
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http://localhost:8888/mystream/index.m3u8
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```
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Although the server can produce HLS with a variety of video and audio codecs (that are listed at the beginning of the README), not all browsers can read all codecs. You can check what codecs your browser can read by visiting this page:
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Although the server can produce HLS with a variety of video and audio codecs (that are listed at the beginning of the README), not all browsers can read all codecs.
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<https://jsfiddle.net/4msrhudv>
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You can check what codecs your browser can read by [using this tool](https://jsfiddle.net/4msrhudv).
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If you want to support most browsers, you can to re-encode the stream by using the H264 and AAC codecs, for instance by using FFmpeg:
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@ -837,6 +907,8 @@ ffmpeg -i rtsp://original-source \
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-f rtsp rtsp://localhost:8554/mystream
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```
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Known clients that can read with HLS are [FFmpeg](#ffmpeg-1), [Gstreamer](#gstreamer-1), [VLC](#vlc) and [web browsers](#web-browsers-1).
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##### LL-HLS
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Low-Latency HLS is a recently standardized variant of the protocol that allows to greatly reduce playback latency. It works by splitting segments into parts, that are served before the segment is complete. LL-HLS is enabled by default. If the stream is not shown correctly, try tuning the hlsPartDuration parameter, for instance:
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@ -1050,7 +1122,7 @@ paths:
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### Save streams on disk
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To save available streams on disk, you can use the `runOnReady` parameter and _FFmpeg_:
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To save available streams on disk, use the `runOnReady` parameter and _FFmpeg_:
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```yml
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paths:
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@ -1409,24 +1481,30 @@ The command will produce tarballs in folder `binaries/`.
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## Standards
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* RTSP
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* [RTSP / RTP / RTCP standards](https://github.com/bluenviron/gortsplib#standards)
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* HLS
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* [HLS standards](https://github.com/bluenviron/gohlslib#standards)
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* RTMP
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* [RTMP](https://rtmp.veriskope.com/pdf/rtmp_specification_1.0.pdf)
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* [Enhanced RTMP](https://raw.githubusercontent.com/veovera/enhanced-rtmp/main/enhanced-rtmp-v1.pdf)
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* WebRTC
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* [WebRTC: Real-Time Communication in Browsers](https://www.w3.org/TR/webrtc/)
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* [WebRTC HTTP Ingestion Protocol (WHIP)](https://datatracker.ietf.org/doc/draft-ietf-wish-whip/)
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* [WebRTC HTTP Egress Protocol (WHEP)](https://datatracker.ietf.org/doc/draft-murillo-whep/)
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* Video and audio codecs
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* [Codec standards](https://github.com/bluenviron/mediacommon#standards)
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* Other
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* [Golang project layout](https://github.com/golang-standards/project-layout)
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## Related projects
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* [gortsplib (RTSP library used internally)](https://github.com/bluenviron/gortsplib)
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* [gohlslib (HLS library used internally)](https://github.com/bluenviron/gohlslib)
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* [mediacommon (codecs and formats library used internally)](https://github.com/bluenviron/mediacommon)
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* [datarhei/gosrt (SRT library used internally)](https://github.com/datarhei/gosrt)
|
||||
* [pion/webrtc (WebRTC library used internally)](https://github.com/pion/webrtc)
|
||||
* [pion/sdp (SDP library used internally)](https://github.com/pion/sdp)
|
||||
* [pion/rtp (RTP library used internally)](https://github.com/pion/rtp)
|
||||
|
|
|
|||
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