I am building a webphone application with jssip. I know that webrtc supports opus and pstn is ulaw and alaw. Does webrtc also support ulaw and alaw? I want to prevent transcoding to optimize CPU usage on the asterisk server.
The ulaw codec is supported in WebRTC, as well as g722 generally. You can just enable it, try a call, and see if it works. If it doesn’t then the call would fail.
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