Yealink Forums

Full Version: Format of Multicast RTP stream
You're currently viewing a stripped down version of our content. View the full version with proper formatting.
Hi

I'm trying to output a pre-recorded wav file to Yealink phones via vlc and RTP Multicast and am experiencing choppy audio.

Phones are T38G's and T42G's configured for PCMU.

When not using vlc and originating the stream from a Yealink phone, multicast from a phone to other phones gives a crystal clear stream.

Using vlc I am transcoding as follows:

--sout "#transcode{acodec=ulaw,ab=16,samplerate=8000,channels=1}:rtp{dst=224.0.1.201,por​t-audio=5439}"

The audio plays but is very choppy.

I'm at a loss as to what I am doing wrong and am questioning the format of the audio stream.

Any help would be much appreciated.

Thanks

Steve
Hi Steve,
Your format :
--sout "#transcode{acodec=ulaw,ab=16,samplerate=8000,channels=1}:rtp{dst=224.0.1.201,por​​t-audio=5439}"
When chennels=1 , did it mean "stereo"? If yes, please change it to Mono which should be " 0".
If the sound is still not stable, please send us the pcap trace that we can check.

Best Regards!
Flora
I ran into the same choppy audio when streaming from PulseAudio in Linux. After a whole lot of fighting and packet tracing I finally came to the conclusion that the packets being sent out were too big for the Yealink phones to handle properly. I set the MTU to 300 instead of the default and instantly everything was clear no more problems.

Not sure how you do it with vlc, but something like --mtu=300 would seem to be correct to add to your command line somewhere:
https://forum.videolan.org/viewtopic.php?t=95549


(05-24-2015 12:29 AM)srcurtis Wrote: [ -> ]Hi

I'm trying to output a pre-recorded wav file to Yealink phones via vlc and RTP Multicast and am experiencing choppy audio.

Phones are T38G's and T42G's configured for PCMU.

When not using vlc and originating the stream from a Yealink phone, multicast from a phone to other phones gives a crystal clear stream.

Using vlc I am transcoding as follows:

--sout "#transcode{acodec=ulaw,ab=16,samplerate=8000,channels=1}:rtp{dst=224.0.1.201,por​t-audio=5439}"

The audio plays but is very choppy.

I'm at a loss as to what I am doing wrong and am questioning the format of the audio stream.

Any help would be much appreciated.

Thanks

Steve
I was struggling to get VLC to work, and was reading about some known bugs in VLC, so I just decided to stop trying to solve bugs Wink

ffmpeg by itself works just fine for me using the following:

ffmpeg -re -i foo.wav -filter_complex 'aresample=8000,asetnsamples=n=160' -ar 8000 -f mulaw -f rtp rtp://234.3.2.1:2345

Enjoy your distributed speaker system!
(07-16-2015 03:53 AM)bnelson Wrote: [ -> ]I was struggling to get VLC to work, and was reading about some known bugs in VLC, so I just decided to stop trying to solve bugs Wink

ffmpeg by itself works just fine for me using the following:

ffmpeg -re -i foo.wav -filter_complex 'aresample=8000,asetnsamples=n=160' -ar 8000 -f mulaw -f rtp rtp://234.3.2.1:2345

Enjoy your distributed speaker system!

I have tried your example unsuccessfully. I have also followed the following wiki without any success as well http://wiki.snom.com/Category:HowTo:Multicast_Audio

Any clue if this will work with a static build of ffmpeg?

I tried the following build:

http://johnvansickle.com/ffmpeg/
If it helps at all, here's my ffmpeg version. Looks a little older than what you were using:

ffmpeg version 2.7 Copyright © 2000-2015 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)

I did have to ensure that the input wav file was already at "phone quality", as full bitrate wav files were also not working. When the input file was "bad", I did still get audio streamed, it just sounded terrible.
Works great. I did a YouTube using this method though I haven't got the audio quality 100%. https://youtu.be/BX12aP5DXuk
Reference URL's