[asterisk-users] RTP handling

[Jeff LaCoursiere]

Hello,

I am wondering if asterisk does anything at all to RTP packets passed 
from channel to channel if no transcoding is involved? Can I assume that 
the packet that left phone A, arrived at the asterisk server, was copied 
to phone B's channel and eventually arrived at phone B had exactly (byte 
for byte) the same payload?  Assume two SIP endpoints, no NAT involved.

Thanks,

j