ossrs / srs

SRS is a simple, high-efficiency, real-time media server supporting RTMP, WebRTC, HLS, HTTP-FLV, HTTP-TS, SRT, MPEG-DASH, and GB28181.
https://ossrs.io
MIT License
25.72k stars 5.39k forks source link

srs As the playing time gets longer, the delay also gets longer---help needed #606

Closed ghost closed 8 years ago

ghost commented 8 years ago

Hi, the environment is as follows: One origin server and one edge server. The broadcaster pushes the stream to the origin server, and the edge server pulls the stream from the origin. We used sb_rtmp_load_fast to stress test 500 clients to the edge server, while also opening a webpage window to play the stream from the edge server. The bandwidth of the edge server only reached 50M/s, even though it has a gigabit network card. Initially, the delay was around 1.2 seconds, but after 30 minutes, it reached 5 seconds. After one hour, it reached a delay of 39 seconds.

I would like to ask, what could be the problem? The player used is http://ossrs.net/srs.release/trunk/research/players/srs_player.html?server=192.168.116.139&vhost=192.168.116.139#.

TRANS_BY_GPT3

ghmack commented 8 years ago

RTMP does not have a time synchronization verification mechanism, so any program, system, or network-related lag will result in increasing delays. To achieve minimal lag, optimization should be done from both the pusher and receiver ends.

TRANS_BY_GPT3

winlinvip commented 8 years ago

What is the bandwidth speed to the edge of the player? Also, you have to keep watching the streamer continuously, you can't switch to other pages.

TRANS_BY_GPT3

ghost commented 8 years ago

hi,播放器到边缘的带宽测速是百兆宽带。。主播就是我自己,,主播和播放器在同一台机器上。。。。我用手机的秒表测试出来的。求答案

winlinvip commented 8 years ago

Try changing the server and network, I have never heard of this issue before, I used to play for a long time without encountering it.

TRANS_BY_GPT3

liutao6982 commented 8 years ago

If there is network jitter, it can indeed lead to buffering and thus cause an increase in latency. However, it is usually the client that experiences the buffering.

TRANS_BY_GPT3

wanghuan578 commented 8 years ago

Enabling gof_cache will solve the issue of delayed accumulation.

TRANS_BY_GPT3

LF-DevJourney commented 7 years ago

I use ffplay with the nobuffer setting ffplay -fflags nobuffer rtmp://192.168.1.66/myapp/live, but there is still an issue of accumulating delay. Initially, there is less than a second of delay, but after half an hour, it increases to over 2 seconds. If I stop ffplay and execute the above command again, the accumulated delay disappears, but it reappears after a while.

TRANS_BY_GPT3

watchpoints commented 4 years ago

How to solve it?

TRANS_BY_GPT3

winlinvip commented 3 years ago

Client problem.

TRANS_BY_GPT3