Playback audio between servers with a specific codec

Hi there,

I´m new to asterisk and have installed the latest certified version to use in a research project I´m conducting.
Basically Im trying to play sample .WAV files on a call from Asterisk server A to a recipient Asterisk server B (both just running on local machines) where they will be recorded. This is to produce clean reference samples (from server A) and degraded samples (recorded from server B) which will be compared to assess Voice Quality.

Im using Asterisks SIP functionality and the EVS codec for asterisk provided by Traud ( to test voice quality for audio encoded by EVS and by OPUS. Ive read about asterisks playback functionality and call initiation commands but I was wondering how I could use these to select the codec for the call and also how to record the call at Server Bs end?

So far Ive set up Asterisk and patched it with the EVS codec as per the instructions so I should be ready to go.

Any help would be greatly appreciated.


You will need a format module for EVS as well as a codec module.

If you limit the codec to only one codec in the SIP configuration files, that codec will be used. If the media is only available in one format, that format will be used.

Be careful with originate, as you can end up with signed linear as the codec and conversions both sides.

Having said that, our setup sounds overcomplicated. If you want to test the result of transcoding, you can do it offline, with no call needed.

Also, in a production environment, you can have the audio in multiple formats, so that transcoding is not needed, or only used in simple cases.

It appears that EVS may be patent encumbered, so you may be limited to non-production use and unable to redistribute the combined software…

Thanks for the quick reply, I’ll edit the sip conf file and see what the result is. I think the format issue was dealt with in the version of Evs I picked up but I’ll reread the documentation.

This is an academic research project so it’s non commercial and I’m not too worried about patenting issues. As for my setup, I’m not just looking to test encoding issues. I’m also simulating network impairment factors using WANem to create the degraded files. Forgot to mention that sorry