Webrtc delay from a rtmp live stream encoded using FFmpeg

[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 November 22, 2022 by ThibaultOber

[ThibaultOber]

When broadcasting with obs everything works as espected, the delay is very small.

This line is not present with obs but does appear with ffmpeg:

2022-11-22 09:22:58,065 [NioProcessor-3] WARN  o.r.s.net.rtmp.InboundHandshake - Client not compatible

 

[ThibaultOber]

It seems the delay came from the audio context.

If we remove the audio context, it works as expected.

Do you have a documentation regarding the AAC configuration and pitfall that can arise ?

Thanks you.

[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.

Edited November 22, 2022 by usama

[usama]

you can refer this

ffmpeg -re -i 'video.mp4'  -c:v libx264 -b:v 1600k -preset ultrafast -b 900k -c:a aac -b:a 128k -s 1920x1080 -x264opts keyint=50 -g 25 -pix_fmt yuv420p -f flv rtmp://stream.serverok.in/LiveApp/225873992721868053400782

[ThibaultOber]

We are linking or program with ffmpeg library, no command line are available 😂.

We have to work using libavformat, libavcodec, ...

We will try to switch directly with opus.

 

Thank you.

[usama]

yes I understood but what I meant was setting configuration similar to that command line with ffmpeg library APIs