Fwd: How to get real time audio streams

Using the ARI, I have been able to create a statis application where I get a bridge, add a snoop channel to it and process the audio over a websocket.

My issue is that I specify the format for my external media as ulaw but when I decode the data obtained using ffmpeg at a rate of 8000, I get very robotic audio which lags.

What am I doing wrong?

@shamnusln should have resolved this I think. Would be glad if you can assist me with this.


It might be helpful to post some actual code you are trying, your ffmpeg options, etc.

Also, if you can confirm the audio you are feeding in to Asterisk is of high-quality, such as by Record() application to a file – which you analyze in something like Audacity – or modify your code to save to file instead of running ffmpeg – then you can probably narrow-down the issue a little more to either an internal or external source.

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