Files
rtsp-simple-server/internal
t-animal 8568d8c57c webrtc: fix random absolute timestamps with Opus, G711 and LPCM (#5597)
When rewriting audio RTP timestamps in WebRTC egress, NTP was
derived using regenerated packet timestamps minus the incoming
RTP base timestamp.

That mixed timestamp domains and could shift absolute time by an
arbitrary offset while still exposing mapping as available.

Fix by using a consistent outgoing RTP domain in rewritten audio
paths:
- snapshot outgoing base timestamp before rewriting each unit
- compute NTP from (outgoing packet timestamp - outgoing base
  timestamp)

This keeps RTP<->NTP mapping coherent for sender reports and prevents
random absolute-time offsets in WebRTC loopback with
useAbsoluteTimestamp.

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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2026-03-21 19:46:39 +01:00
..
2026-03-21 18:22:49 +01:00
2026-03-17 17:22:15 +01:00
2026-03-17 17:22:15 +01:00
2026-02-21 14:48:53 +01:00