Hi Ross, Yes, for function RTPInterface::sendDataOverTCP , I found it will not resend the header data if sent failed for TCP, then it will lose the packet with this header. We need to check why data sent failed for our OS. Thanks for your response. :)
Thanks. /Qian -----邮件原件----- 发件人: live-devel <live-devel-boun...@us.live555.com> 代表 Ross Finlayson 发送时间: 2020年7月29日 17:14 收件人: LIVE555 Streaming Media - development & use <live-de...@us.live555.com> 主题: Re: [Live-devel] [live555] RTP sequence number is not continuous when video bitrate is high (for example: 8Mbps) > On Jul 29, 2020, at 9:05 PM, Zhang Qian（张倩） <qianzh...@asrmicro.com> wrote: > > > Hi Ross, > > > Sorry for misunderstanding. I catch the tcpdump log for rtsp server, and I > use the TCP socket. Seems that these FU packets are not sent to TCP protocol > stack. So I want to check whether these packets are lost in rtsp server for > large bitrate. Your stream’s bitrate is exceeding the capacity of your network. You will lose RTP packets, ***even if you are streaming RTP-over-TCP***. (If you are streaming RTP-over-TCP, then eventually the TCP socket, inside the sender’s OS, will run out of buffer space, and several writes to the TCP socket - by the RTSP server - will fail. This is what you are seeing.) You CANNOT avoid data loss if your stream’s bitrate exceeds the capacity of your network (which is usually a lot lower than the nominal ‘speed’ of your network interface). This is physically impossible. The ONLY solution is to reduce your stream’s bitrate. Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel _______________________________________________ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
