[asterisk-users] RTP traffic through Asterisk??

[Ignacio]

I have just established a call between 2 sip phones and I have noticed
that all RTP traffic goes through Asterisk Server.

I was expecting RTP traffic went to one phone to another phone directly.

I set canreinvite=yes in sip.conf in both sip peers.

I also tested it with 2 mgcp phones and same result, all rtp traffic
goes through Asterisk.

Is there any way to force traffic to go from one phone to another?

Thank you very much.