[Live-devel] minimum inter-packet delay

[Ross Finlayson]

>I have a linux application send uncompressed frames using the liveMedia
>library; with wireshark I'm measuring 17 microseconds between successive
>packets (sized 1490 bytes, of the same image frame) being sent, which
>boils down to instantaneous bandwidth consumption of 700 Mb/s. In this
>configuration, several packets are lost at some receivers. I've already
>tried increasing the UDP sending and receiving buffers - but this
>doesn't solve the packet loss.
>
>A temporary work-around seems to increase the interpacket gap to 30
>microseconds by inserting a few fprintf()s in sendPacket() in
>RTPInterface.cpp. However, isn't there a cleaner solution to specify the
>minimum inter-packet gap in liveMedia?

Are you setting the "fDurationInMicroseconds" variable correctly in 
whatever source object(s) feed into your "RTPSink" (or 
"BasicUDPSink") object?  It's that variable that determines the 
(average) inter-packet gap.

However, what is your data's bitrate *supposed* to be?  If it really 
is supposed to be 700 Mbps, then you're not going to be able to 
increase the average inter-packet delay - it is what it is.

Note that becuase the "LIVE555 Streaming Media" code is 
single-threaded code that simply reads sequentially from a socket, 
any packet loss that you see cannot be the fault of this code. 
Instead, it must be either actual network packet loss, or data loss 
(e.g., overflowing buffers) within your OS.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/