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https://github.com/aler9/rtsp-simple-server
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40 lines
1.7 KiB
Markdown
40 lines
1.7 KiB
Markdown
# GStreamer
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GStreamer can read a stream from the server in several ways. The recommended one consists in reading with RTSP.
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## GStreamer and 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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```
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For advanced options, read [RTSP-specific features](../2-features/26-rtsp-specific-features.md).
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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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```sh
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gst-launch-1.0 whepsrc whep-endpoint=http://127.0.0.1:8889/stream/whep use-link-headers=true \
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video-caps="application/x-rtp,media=video,encoding-name=H264,payload=127,clock-rate=90000" \
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audio-caps="application/x-rtp,media=audio,encoding-name=PCMU,payload=0,clock-rate=8000" \
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! rtph264depay ! decodebin ! autovideosink
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```
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While the command for reading an audio-only Opus stream is:
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```sh
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gst-launch-1.0 whepsrc whep-endpoint="http://127.0.0.1:8889/stream/whep" use-link-headers=true \
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audio-caps="application/x-rtp,media=audio,encoding-name=OPUS,payload=111,clock-rate=48000,encoding-params=(string)2" \
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! rtpopusdepay ! decodebin ! autoaudiosink
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```
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While the command for reading a H264 and Opus stream is:
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```sh
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gst-launch-1.0 whepsrc whep-endpoint=http://127.0.0.1:8889/stream/whep use-link-headers=true \
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video-caps="application/x-rtp,media=video,encoding-name=H264,payload=127,clock-rate=90000" \
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audio-caps="application/x-rtp,media=audio,encoding-name=OPUS,payload=111,clock-rate=48000,encoding-params=(string)2" \
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! decodebin ! autovideosink
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```
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