I’ve been looking around the community and mostly see jitterbuffer being applied to the read side, I still don’t know what this means as I’m very new to Asterisk, but can’t seem to find good information on how to use it on the write side, I saw that you can use a pre-dial routine to set the jitterbuffer but I don’t really see anything happening or any difference. I’m trying to buffer the audio for about 2-3 seconds before it’s heard on the other side of the call. All that I’ve manage to do is skip the first few seconds of the sound that comes out.
It means audio received from the remote device is put into the jitterbuffer, instead of audio sent.
Any suggestions on how I can buffer the audio before it comes out on the other side?
I’ve answered this once or twice recently. Any suggestions I have would be found on those topics.
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