[asterisk-dev] RTP Header manipulation

[Francesco Mayer]

Hello everyone,
just for prototyping purpose, I need to allow to custom sip agents to
communicate with PTT (push to talk), using the mark field inside the rtp
header, by means of conferences (ConfBridge application) in an Asterisk
scenario.
Unfortunately I discovered that Asterisk replace the rtp header of the
packet coming from the sip agent with a new one built before to route the
rtp packet to destination.
Could someone kindly suggest me the correct path to follow in order to
solve this problem ?

Thanks in advance,
Francesco Mayer
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