What causes delays in Google WebRTC calls?
(https://webrtc.googlesource.com/src/+/refs/heads/master)
A(Android)->B(IOS) delay occurs, B(IOS)->A(Android) no delay.
Please tell me which part of the code I should trace.
Changed to keep audio buffers short.
Is there a way to control the audio stream related to performance during a call?
Thank you.
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