Thank you for your patience.
It's very depressed to heard that there is no a good way to solve my problem.I don't know what number my server buffertime is.But the number should be small. Our video demands to be a real-time live show, so the buffertime need to be very small, 9 second is too big.
I read NetStream API, it says if I set bufferTime=0, the stream is real-time.But I don't understand why the picture is discontinuous, the FPS number is unstable and floating.
I try to set the client bufferTime=9. It takes about 9 second to play normally. I think because my server is taking pictures in real time, so it needs 9 second to fill the client buffer. I test :
- the client(bufferTime>0) plays dozens of second video in a bad network, it will encounter many time WAITING message. Then the network becomes normal, the client bufferLength number is increasing, sometimes reaches 60(s). But the bufferLength will reduce to 1~5 eventually, even though the process takes dozens of minutes. It's very strange to my client. I check the NetStream API, the maxBufferTime is useless when the bufferTime>0. I really hope the maxBufferTime can be useful in this case, but...
- the client's(bufferTime=0) FPS value is float(I still don't know why this happen).And then it plays dozens of second video in a bad network, it will encounter many time WAITING message. Then it displays picture as soon as possible when the network becomes normal. It this type client, the bufferLength value is always samll. I think this client is real-time player, and I want to know why this client picture is discontinuous.
Is there no other way?
> .I don't know what number my server buffertime is
you should know it, or you waste your time to try to solve your issue.
Maybe you are not carefully reading my comments, so I write it again:
If you expect some stream cut to the clients that's because:
- the audio/video quality and frame per sec is too high for the client network bandwidth capacity
- As the network bandwidth is to low the buffertime is not adapted for it.
you said it's a live show, so since there is no interation between the broadcaster and the watcher it doesn't matter there is 0 sec or 30 second latency. Now if you talk about realtime 2 way communication that's another story. it really depends on the network quality of each other, but to be sure it will work in any case, again reduce the audio/video quality and the number of frames per sec
or for sure you will face of lags and cuts.
live audio video is not a small thing, some guys work on it as a full time job.
some tutorial about live streaming with rtmp:
Stream live media (RTMP) with Adobe Media Server
PS: RTMFP is a better live streaming protocol since it tolerate to skip frames and keep latency low.
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