[Live-devel] RTP packet size

[Filippo Cacace]

 I am using the RTSP server for RTP (video) streams on
low bit-rate links (such as cellular data networks). The
problem I am having with the LIVE library is that the dimension
of the packets is too small at low resolution. For example, a stream
at 100 kbps contains many packets with only 20-30 payload bytes.
The header overhead (VPN+IP+UPD+RTP) consumes the scarce bandwidth,
(moreover, I am using IPv6, thus the IP header is 40 bytes...)
since the server sends too many small packets.

If I remember well, VideoLAN has a different timing, as it usually
creates bigger packets; it can also change the packet size when the
delay on the network grows, attempting to mantain the throughput.

Is there a way to modify the LIVE algorithm in charge of deciding
when to send a packet and how much to fill it? 
I have seen that the function
MultiFramedRTPSink::sendPacketIfNecessary() is in charge of
deciding how much to fill the packet. Is it sufficient to make
a change at this level? Which are the relevant parameters 
that decide the whole thing?

Thanks in advance,

Filippo Cacace