Question

I have an VOIP application i'd like to implement, that requires me to process the audio from a call in real time during the call.

Currently, I'm using Asterisk to handle my calls, and it looks like there's a functionality built in called Audiohooks which is designed to let me access the audiostream, and process it from the dialplay

However, I can not find any documentation whatsoever on how to actually create my own audio hook, nor any recent examples on how it should be done. Are there resources that show how I could use this?

Thanks

Was it helpful?

Solution

That api is availible when you do c/c++ modules for asterisk. No external API.

For examples you can check MixMonitor,func_volume,app_conference and other similar application already developed.

Hint: after work done, you have test for memory leaks and hi-load/concurrent load. Code must be thread-safe.

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