Are there tweaks we can use for adaptive playout buffer? there was one test we had where we made a call over a jittery and lossy link, the call ended up with around 5-7 seconds latency. we had rtp relay on via sems. but the audio was fine, it was just late. so i am kinda thinking this might be because of the buffering.
Can we configure it to drop packets if jitter is too high? Kelvin Chua
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