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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