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docs: update (#5115)
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4 changed files with 21 additions and 5 deletions
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@ -524,6 +524,8 @@ d.video_0 ! rtspclientsink location=rtsp://localhost:8554/mystream
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The resulting stream is available in path `/mystream`.
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The resulting stream is available in path `/mystream`.
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For advanced options, see [RTSP-specific features](rtsp-specific-features).
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#### GStreamer and RTMP
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#### GStreamer and RTMP
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```sh
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```sh
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@ -540,6 +542,8 @@ videotestsrc ! video/x-raw,width=1280,height=720,format=I420 ! x264enc speed-pre
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audiotestsrc ! audioconvert ! avenc_aac ! mux.
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audiotestsrc ! audioconvert ! avenc_aac ! mux.
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```
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```
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For advanced options, see [RTSP-specific features](rtsp-specific-features).
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#### GStreamer and WebRTC
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#### GStreamer and WebRTC
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Make sure that GStreamer version is at least 1.22, and that if the codec is H264, the profile is baseline. Use the `whipclientsink` element:
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Make sure that GStreamer version is at least 1.22, and that if the codec is H264, the profile is baseline. Use the `whipclientsink` element:
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@ -207,6 +207,8 @@ GStreamer can read a stream from the server in several way. The recommended one
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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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gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/mystream latency=0 ! decodebin ! autovideosink
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```
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```
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For advanced options, see [RTSP-specific features](rtsp-specific-features).
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#### GStreamer and WebRTC
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#### GStreamer and WebRTC
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GStreamer also supports reading streams with WebRTC/WHEP, although track codecs must be specified in advance through the `video-caps` and `audio-caps` parameters. Furthermore, if audio is not present, `audio-caps` must be set anyway and must point to a PCMU codec. For instance, the command for reading a video-only H264 stream is:
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GStreamer also supports reading streams with WebRTC/WHEP, although track codecs must be specified in advance through the `video-caps` and `audio-caps` parameters. Furthermore, if audio is not present, `audio-caps` must be set anyway and must point to a PCMU codec. For instance, the command for reading a video-only H264 stream is:
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@ -12,12 +12,18 @@ The RTSP protocol supports several underlying transport protocols, that are chos
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To change the transport protocol, you have to tune the configuration of the client you are using to publish or read streams. In most clients, the default transport protocol is UDP.
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To change the transport protocol, you have to tune the configuration of the client you are using to publish or read streams. In most clients, the default transport protocol is UDP.
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For instance, FFmpeg allows to change the transport protocol with the `-rtsp_transport` flag:
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FFmpeg allows to change the transport protocol with the `-rtsp_transport` flag:
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```sh
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```sh
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ffmpeg -rtsp_transport tcp -i rtsp://localhost:8554/mystream -c copy output.mp4
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ffmpeg -rtsp_transport tcp -i rtsp://localhost:8554/mystream -c copy output.mp4
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```
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```
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Available options are:
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- `-rtsp_transport tcp` to pick the TCP transport protocol
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- `-rtsp_transport udp` to pick the UDP transport protocol
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- `-rtsp_transport udp_multicast` to pick the UDP-multicast transport protocol
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GStreamer allows to change the transport protocol with the `protocols` property of `rtspsrc` and `rtspclientsink`:
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GStreamer allows to change the transport protocol with the `protocols` property of `rtspsrc` and `rtspclientsink`:
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```sh
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```sh
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@ -25,6 +31,12 @@ gst-launch-1.0 filesrc location=file.mp4 ! qtdemux name=d \
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d.video_0 ! rtspclientsink location=rtsp://localhost:8554/mystream protocols=tcp
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d.video_0 ! rtspclientsink location=rtsp://localhost:8554/mystream protocols=tcp
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```
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```
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Available options are:
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- `protocols=tcp` to pick the TCP transport protocol
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- `protocols=udp` to pick the UDP transport protocol
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- `protocols=udp-mcast` to pick the UDP-multicast transport protocol
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VLC allows to use the TCP transport protocol through the `--rtsp_tcp` flag:
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VLC allows to use the TCP transport protocol through the `--rtsp_tcp` flag:
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```sh
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```sh
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@ -2,15 +2,13 @@
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## Security of released binaries
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## Security of released binaries
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Binaries published in the [Releases](https://github.com/bluenviron/mediamtx/releases) section of GitHub are the output of a process which has been designed with a security-first approach. Every step from source code to the intended final destination of binaries is fully visible, immune from external interference and can be independently verified.
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Binaries published in the [Releases](https://github.com/bluenviron/mediamtx/releases) section of GitHub are the output of a process which has been designed with a security-first approach. Every step from source code to the intended final destination of binaries is fully visible, immune from external interferences and independently verifiable. This is the process:
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This is the process:
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1. During every release, the [Release workflow](https://github.com/bluenviron/mediamtx/actions/workflows/release.yml) is triggered on GitHub.
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1. During every release, the [Release workflow](https://github.com/bluenviron/mediamtx/actions/workflows/release.yml) is triggered on GitHub.
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2. The release workflow pulls the source code and builds binaries.
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2. The release workflow pulls the source code and builds binaries.
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3. The release workflow computes SHA256 checksums of binaries and publishes them on the Sigstore Public Good Instance through [GitHub Attestations](https://docs.github.com/en/actions/concepts/security/artifact-attestations).
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3. The release workflow computes SHA256 checksums of binaries and publishes them to the Sigstore Public Good Instance through [GitHub Attestations](https://docs.github.com/en/actions/concepts/security/artifact-attestations).
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4. Checksums and binaries are published on the Release page.
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4. Checksums and binaries are published on the Release page.
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