I am currently running a c++ application which emit a rtmp live stream using FFmpeg and libopenh264 as the encoder. This rmtp live stream then pass through antmedia server which convert it to a webrtc live stream.
- reduce the Iframe interval with the gop parameter to 1 every five frame (gop_size = 5)
- reduce the probe size parameter to 32 (Maximum number of bytes read from input in order to determine stream properties.)
- reduce the maxAnalyseDuration to 32 (Maximum duration (in AV_TIME_BASE units) of the data read from input inavformat_find_stream_info())
I am still noticing a 12 seconds delay inside the html viewer from the antmedia interface. When I am reading the log I saw that antmedia are reencoding the stream using libopenh264.
Is there some official guidelines to reduce delay from a rtmp live stream to a webrtc application ?
Here are the logs from the antmedia platform also attach in log format.
I saw the gopsize getting updated to 60fps, so the video is getting re-encoded. Is there a way to stop this process.
Latency is a huge deal in our application, thank you for your time.
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ThibaultOber
Hello,
I am currently running a c++ application which emit a rtmp live stream using FFmpeg and libopenh264 as the encoder. This rmtp live stream then pass through antmedia server which convert it to a webrtc live stream.
I am following this guide to reduce delay: Minimizing the initial delay | VideoKit (iosvideokit.com)
I set those specific parameters:
- reduce the Iframe interval with the gop parameter to 1 every five frame (gop_size = 5)
- reduce the probe size parameter to 32 (Maximum number of bytes read from input in order to determine stream properties.)
- reduce the maxAnalyseDuration to 32 (Maximum duration (in AV_TIME_BASE units) of the data read from input in avformat_find_stream_info())
I am still noticing a 12 seconds delay inside the html viewer from the antmedia interface. When I am reading the log I saw that antmedia are reencoding the stream using libopenh264.
Is there some official guidelines to reduce delay from a rtmp live stream to a webrtc application ?
Here are the logs from the antmedia platform also attach in log format.
I saw the gopsize getting updated to 60fps, so the video is getting re-encoded. Is there a way to stop this process.
Latency is a huge deal in our application, thank you for your time.
ant-media-server.log
Edited by ThibaultOberLink to comment
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usama
Webrtc does not support ACC audio encoding so AMS transcode ACC to Opus. Can you plese try setting the key frame interval to once every 2 sec.can you try setting the encoder speed as veryfast.
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