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fix
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@ -39,10 +39,10 @@ go2rtc:
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streams:
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test_cam:
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- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio.
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- "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
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- "ffmpeg:test_cam#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
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test_cam_sub:
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- rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio.
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- "ffmpeg:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
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- "ffmpeg:test_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
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cameras:
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test_cam:
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