Closed linkewei0580 closed 1 year ago
The rtc_to_rtmp function relies on the SR (Sender Report) in the RTCP protocol. Could you capture packets to check if OBS is sending RTCP?
Could you help me take a look? Client address: 192.168.1.107, SRS address: 192.168.1.123
TRANS_BY_GPT3
Description
obs whip streaming
dvr does not take effect, file size is 13 bytes.
Enabling rtc_to_rtmp on cannot play flv.
SRS Version: SRS/6.0.51(Bee), MIT
conf
hls { enabled on; }
http_remux { enabled on; mount [vhost]/[app]/[stream].flv; } rtc { enabled on;
@see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
}
[2023-06-16 11:01:28.031][INFO][30727][31a7l155] dvr stream livestream2 to file /home/install/vidoes/live/livestream2.1686884488030.flv [2023-06-16 11:01:28.031][INFO][30727][31a7l155] ignore disabled exec for vhost=defaultVhost [2023-06-16 11:01:28.031][INFO][30727][7813w9kq] RTC: Need PLI ssrc=5000, play=[7813w9kq], publish=[31a7l155], count=1/1 [2023-06-16 11:01:28.031][INFO][30727][7813w9kq] RTC: Need PLI ssrc=5002, play=[7813w9kq], publish=[31a7l155], count=1/2 [2023-06-16 11:01:28.031][INFO][30727][31a7l155] RTC: Request PLI ssrc=5000, play=[7813w9kq], count=1/1, bytes=12B [2023-06-16 11:01:28.031][INFO][30727][31a7l155] RTC: Request PLI ssrc=5002, play=[7813w9kq], count=1/2, bytes=12B [2023-06-16 11:01:29.058][INFO][30727][7813w9kq] update source_id=31a7l155/f0a340p2 [2023-06-16 11:01:29.059][INFO][30727][7813w9kq] RTC: Jitter rebase value=0, last=3242, distance=3242, pkt-base=0/0, correct-base=53523/56765 [2023-06-16 11:01:29.061][INFO][30727][7813w9kq] RTC: Jitter rebase value=0, last=3012418652, distance=-1282548644, pkt-base=2525107968/0, correct-base=222625516/709936200