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WebRTC: Support G711A audio codec #4075
Comments
SRS only support srs/trunk/src/app/srs_app_rtc_conn.cpp Lines 2661 to 2664 in 282d94d
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I think it's reasonable to support G711 codec with WebRTC. Note that SRS will convert Opus/G711 to AAC for RTMP/HTTP-FLV/HLS if enabled converting RTC to RTMP. |
The first problem is prepare a test/dev env, how to prepare a webrtc client with G711 audio codec? The common WHIP request sent from the web browser, which generate the sdp offer with @thanhbinh89 Could you describe how your webrtc env with G711 works? |
I am using a G711 sample file, which has been recorded from a camera device.
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@thanhbinh89 How do you publish above G711 files to SRS by RTC? more details step to step? |
@suzp1984
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Describe the bug
Received a 502 Bad Gateway response when sending the WHIP. The SDP content in the body of the WHIP is provided below. The SDP includes video (H.264) and audio (G.711a), not Opus.
Version
Docker ossrs/srs:5 , ossrs/srs:6
To Reproduce
sdp
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