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VERSION:2.0
PRODID:-//Allsikt//Article Deadline//EN
CALSCALE:GREGORIAN
METHOD:PUBLISH
BEGIN:VEVENT
UID:e03c413c184354789850b32a0a823c8b5a6c0c06@allsikt.tech
DTSTAMP:20260728T104737Z
DTSTART;VALUE=DATE:20260728
DTEND;VALUE=DATE:20260729
SUMMARY:WebRTC Aggressively Drops Audio Packets to Keep Latency Low
DESCRIPTION:Check your WebRTC implementation for packet drops\; add a 200 ms buffer to improve audio quality.\n\nSource: Simon Willison\nOpen: https://www.allsikt.se/article/webrtc-aggressively-drops-audio-packets-to-keep-latency-low-5ce10b00
URL:https://www.allsikt.se/article/webrtc-aggressively-drops-audio-packets-to-keep-latency-low-5ce10b00
STATUS:CONFIRMED
TRANSP:TRANSPARENT
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