Hi all,
I would like to make asterisk acting like a “man in the middle” for some tests with ZRTP module (http://zfoneproject.com/prod_asterisk.html ). I need to implement the topology described in figure 1.2.3 ZRTP call in MiTM mode for non-ZRTP user from the doc : http://zfoneproject.com/docs/asterisk/man/html/u_guide.html#sessions_mitm
However MITM scenario does not work unless i change the codecs config on sip clients to force asterisk to transcode the voice. I believe this is not a problem with ZRTP, but just with asterisk configuration.
So my question is:
How could I force asterisk to process RTP traffic except when codec translation is needed ?
I already configured sip.conf with the canreinvite=no (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite) to make all RTP traffic flow through asterisk, but i believe that asterisk just forwards the traffic, cause the MITM does not work without the use of transcoding.
Sorry for all the links, i hope i made the point clear and thank you in advance for taking the time to read this !