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Support G711 in RTMP #2857
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aler9
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aler9
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This issue is mentioned in release v1.4.2 🚀 |
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Describe the feature
Hello,
The idea is support G711 in RTMP protocol because currently you have no way without re-encoding to reproduce audio using WebRTC if the input is provided by a RTMP stream in this server.
This allow users to use one of the most commons protocols like RTMP and reproduce the audio in a browser with low latency using WebRTC without re-encoding.
Ffmpeg command to test:
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