[Live-devel] Slow connection problem with RTP over RTSP/TCP
Ross Finlayson
finlayson at live555.com
Wed Nov 18 07:49:58 PST 2009
No, this is nonsense. TCP is intended to be a reliable byte-stream
protocol; it's the job of the operating system's TCP implementation -
not application-level code (such as LIVE555) - to provide reliable
delivery. If, however, your stream's bitrate is too large for your
network, then you're inevitably going to get packet loss (usually
because an OS buffer in the sender OS will overflow). There's
nothing you can do to avoid this. If your stream's bitrate is really
too large for your network, then you should not be sending it (and
you should certainly not be sending it over TCP).
If, however, your stream's *average* bitrate is within the capacity
of your network, you can reduce the probability of your server OS's
TCP implementation losing data by increasing its OS socket buffer.
Note that we provide a function "increaseSendBufferTo()" that does
this. Our RTSP server implementation - by default - sets this buffer
size (for each server and stream socket) to 50 kBytes (search for
"increaseSendBufferTo" in the code). However, you could use a larger
value this (e.g., by subclassing "RTSPServer").
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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