[asterisk-users] RTP traffic through Asterisk??

[Ignacio]

Ok, thank you very much. I will try to find any information in
asterisk documentation about RTP.

On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III
<jsullivan at opensourcedevel.com> wrote:
> 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
>
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
>
>
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