Hi ibc,
When i use ffmpeg to do h264 stream rtp-packetize and push to mediasoup server for a realtime streaming scenario, i see the repeated logs:
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:10064] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:10368] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:10617] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11071] +2s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11249] +867ms
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11390] +879ms
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11587] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11761] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:11960] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:12326] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:12683] +2s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:13035] +2s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:13167] +971ms
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:13358] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:13598] +935ms
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:13956] +1s
mediasoup:Channel [pid:3549] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:14478] +2s
i thought ffmpeg does not do RTCP PLI feedback from source, (it may do SR, because there is in ffmpeg’s code), while checking libwebrtc’s impl, libwebrtc seems do encode keyframe realtime to response to RTCP PLI request.
does it affect the webrtc end2end latency? The input source is assumed to be realtime-encoded h264 stream, which doesn’t do any webrtc/rtp-related things.