Audio quality

[billybigpotatoe]

We are streaming conference calls from a PBX to the web using the FMG / FMS combination to convert the calls from SIP to RTMP. The RTMP stream is being passed to a CDN for delivery to the audience.

Our current deployement is resulting in very clipped / tinny audio - we have tried many configs on both the RTMP, Speex side and we have also tried serveral different Codecs (NM8k, NM22k and Speex) and volume options.

Please can someone advise regarding settings to produce the best quality audio.

We can output our SIP stream using the Speex codec - would it be possible to remove the need for any transcoding by FMG?

As we are passing the stream to a CDN we are doing all the processing in FMS e.g.

application.legService.onLeg = function(info) {

  application.legService.setWriteCodec(info.legID, "speex");

  application.legService.setWriteAudioGain(info.legID, 70);

  application.legService.setLegStateSendRecv(info.legID);

  return true;

}

All advice greatly appreciated.

D.

[Guest]

Hello,

         Getting flawless audio quality is definitly easy to achieve.

         Try out following steps: 

              1) Stick to codec nm22k; this provides by far the best output quality with FMG.

              2) In rtmp.xml; set <SilenceLevel> to 0 (inside <audio> tag)

              3) Since your use case in only one way broadcast, On flash player set NetStream.bufferTime  to 0.1 (a non zero value); this will make sure that flash player isn't dropping audio packets. Usually, it is not required, but if client bandwidth is really choppy; this will help ensure quality.

              4) Check in FMG core00.log; does FMG log errors indicating frequent packet drops ?

         I hope it helps

-Pankaj

[billybigpotatoe]

Do you have any recomendations regarding codecs to use between SIP and FMG?

D.

[Guest]

Default codec pre-configured in sample sip.xml FMG sounds perfect (i.e. G711u); so no special reccomendations on codec between sip & FMG.

[billybigpotatoe]

We have been running more tests - we are not seeing any dropped frames on the player side or in the server logs.  The audio appears to be stable but the output sounds very harsh - especially at the top end. 

I am happy to run a demo session with you - so you can hear the issue. Our current transocde (based on flash client software) path still is sounding better so unless I can at least match the audio quality I will need to investigate alternative solutions.

Are there any support options for this product beyond this forum?

[Guest]

Excessive gain causes harsh audio. Default audio gain is just fine in most cases. Try removing following call from your custom app:

  application.legService.setWriteAudioGain(info.legID, 70);

[billybigpotatoe]

Audio still the same - just quieter! Any other tips - I am nervous that we may be aiming for different quality levels.  The audio is working - we are not hearing clipping or seeing dropped packets - it is more the sound of artifiacts - from resampling. I see Speex has an option of a filter - certainly filtering the top end of the audio would I expect fix the issue - alternatively ensuring that there is no transcoding / resampling going on.

Any thoughts?

[Guest]

To confirm, are you using G711u (default codec entry) in sip <profile> in sip.xml ?