[Live-devel] 答复: [live555] RTP sequence number is not continuous when video bitrate is high (for example: 8Mbps)

[Zhang Qian（张倩）]

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.


Thanks.
/Qian

-----邮件原件-----
发件人: live-devel <live-devel-bounces at us.live555.com> 代表 Ross Finlayson
发送时间: 2020年7月29日 16:45
收件人: LIVE555 Streaming Media - development & use <live-devel at 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 8:34 PM, Zhang Qian（张倩） <qianzhang at asrmicro.com> wrote:
> 
>  
> Hi Ross,
>  
> I found sometimes RTP sequence is not continuous when video bitrate is very high such as 8Mbps. Seems that it will lose some FU packets.

Congratulations!  You have discovered datagrams!  UDP (and RTP, which uses UDP) is an unreliable datagram protocol.  Packets can get lost (especially if you’re streaming at a high bitrate - approaching the capacity of your network).


> How to avoid it.

You can’t.  However, you can usually reduce your packet loss rate by streaming at a lower bit rate (which requires lowering your encoder’s resolution and/or frame rate).

If you don’t want any data loss at all, then you shouldn’t be doing real-time video streaming.  Instead, transfer your video files using TCP (e.g., using HTTP).  Or else use something like HLS - where you get no data loss, but at the cost of high (non-real-time) latency.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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