how to reduce latency when play rtp stream

[Soho Soho123]

Hi All,

I create a pipeline,
appsrc-->decodebin2-->audioconvert-->audioresample-->alsasink,
the src is from rtp socket that created by myself.
The latency is very long when play rtp stream.
Does anyone have idea how to reduce latency between rtp socket input
and alsa output?
Does anyone have idea about how to set the min latency and max latency
property of appsrc?
Which value is suitble for reducing latency?
And if I reduce max-bytes, does that helpful for reducing latency?

Thanks!
Soho