[asterisk-users] RTP traffic through Asterisk??

[John A. Sullivan III]

On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:
> 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?
<snip>
I don't recall where it is off-hand but, somewhere in the Asterisk
documentation, there is an explanation of how Asterisk makes a decision
about reinvites.  You may want to look at that to see if your
environment satisfies all the requirements and how it can be adapted if
it does not - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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