I wonder if anyone can offer any insights... We are using OpenSIPS 3.1 as a mid-registrar and in front of an Asterisk box. We include incoming WebRTC traffic using the OPUS codec. Which do you think would be the better option:
1 - Pass OPUS directly through to Asterisk 2 - Use RTPEngine to transcode OPUS to PCMU before passing it on to Asterisk to reduce the workload on the Asterisk box If option 2 would be the more efficient option, are there any settings we should consider to allow transcoding to be as efficient as possible?
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