[Live-devel] RTP over TCP packet size determination

[Lan Zang]

Hi,

I sent the audio and video muxed in MPEG2TS container to an mobile phone(Android) via live555. The traffic from Live555 server to phone is via RTP over TCP, for the mobile phone is behind a NAT gateway. It goes well until I was request to send audio data only. Because that the bitrate for audio only stream is rather lower, the receiver running on mobile phone meets a jittering of the RTP/TCP stream as large as 3 seconds. I found if I split the packet into rather smaller size, the receiver's jittering situation is improved.

So, I just want to see if I can change some parameters inside the server, so, the Live555 can send out smaller size packets? I found there is line47 in MultipleFramedSink.cpp:
setPacketSizes(1000, 1448);

If I changed the value to smaller one, like (256, 400), the packet size keeps the same size(~1300). 

Regards,
Lan Zang(Sander)