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versatica / mediasoup

Cutting Edge WebRTC Video Conferencing
https://mediasoup.org
ISC License
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TestRtpStreamSend: ASAN, fix buffer overflow #1419

Closed jmillan closed 4 months ago

jmillan commented 4 months ago

Do not use a hardcoded length value when parsing a RTP packet

Fixes TODO 1 in https://github.com/versatica/mediasoup/issues/1417#issuecomment-2199943723