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Published 20 hours ago • verified publisher icon ossrs

WebRTC: Add audio jitter buffer in rtc2rtmp. Increase the jitter buffer for audio.

Open xiaozhihong opened this issue 2 years ago • 3 comments

The audio RTP packet receives in WebRTC publisher transcodes from OPUS to AAC directly when enabled rtc_to_rtmp option. But it may be out of order or arrive after retransmitting, so we need an audio jitter buffer to make it in order with minimal latency.

TRANS_BY_GPT3

xiaozhihong avatar Mar 07 '23 11:03 xiaozhihong

If out of order, the audio stream will be corrupt?

winlinvip avatar Mar 11 '23 03:03 winlinvip

@winlinvip can you add, please? this is a real problem...

green-cats avatar Aug 15 '24 23:08 green-cats

Patch is welcome.

winlinvip avatar Aug 15 '24 23:08 winlinvip

Supported in SRS 7.

winlinvip avatar Oct 30 '25 11:10 winlinvip