[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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