Can I tune the size of internal buffer, when I make live stream for rtsp stream player? If want increase this size, what must I change?
Why do you think you want to increase this size. What are you trying to accomplish. Please provide more detail.
Not sure I follow the logic. How is thing going to help?
The buffering should be done client side (in the player) to smooth playback. The server-side buffering does not directly affect playback. For example in Flash you can set the NetSTream.bufferTime to control the size of the client-side buffer.
If you feel you need a buffer on the edge, then the network connection between the origin and edge(s) is not adequate. Still, the issue is probably between the edge and the client, and standard practice of buffer on the client can help some, but streaming at an appropriate bitrate per client connection is the best solution. Use multi-bitrate streaming to accomplish that, see that section in the live tutorial.
And what about Wowza Repeater? We try to send stream to large distance and sometimes we have lags too. Origin server is set as “live” and repeater as “live-repeater-edge”. We want to try large buffer on edge server to get rid of lags.
The final goal is sending stream as RTP to large distance. We try to send it through RTMP for guaranteed delivering, and at the end point we will push RTP MPEG2-TS WRAPPED out from end point edge server.
I’m using external transcoder program to produce live stream. Sometime I have lags – I think that if I enlarge buffer, I get rid of them.
Maybe I don’t understand something, bat it is common technique to produce smooth output – dampering input with large enough buffer. Or not?