HI
 
I have a VoIP system up work working quite well on the office network.  When I try to run the program across a couple of modems (33.6k) the audio seems to break up a lot making it un-useable, I'm putting this down to the bandwidth.
 
Even with the bandwidth problem I think this should work better than it does.  The audio is compressed with GSM 6.10, it tries to send 5 packets of data each around 350 bytes for 1 seconds worth of audio.
 
I'm wondering if there is some delay with WinSock and UDP that I can alter to fix this or if there is anything else I missed??
For TCP, there's the Nagle's algorithm, which causes the TCP layer to collect data to an extent before sending it across the wire so that data transfers would be more efficient (the overhead, including TCP/IP headers and CPU time, per bytes of data gets minimized.)  I'm not sure if this algorithm works for UDP or not, but if it does, I think you need to disable it to get a "constant" data transfer rate.
 
Long time ago when I tried VoIP on my dial up connection, it was perfectly feasible to actually talk to someone...
And if great things have been a failure with you, have you yourselves therefore- been a failure? And if you yourselves have been a failure, has man therefore- been a failure? If man, however, has been a failure: well then! never mind!
-Thus Spoke Zarathustra, F. W. Nietzsche
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