[asterisk-users] RTP directly

Benjamin Jacob benjamin.jacob at mgl.com
Wed Jan 10 21:32:16 MST 2007


Davida,
You would also want to look at canreinvite option in sip.conf
http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite

cheerz
- Ben.


Eric "ManxPower" Wieling wrote:

> David Alcott wrote:
>
>>
>> Is there a way to configure the Asterisk so that the RTP goes 
>> directly between the Endpoints as opposed to going through the asterisk?
>
>
> That is the default if Asterisk believes it will work.  Things that 
> might not make it work is tTwW options to Dial, protocol transation 
> (one leg is SIP, the other is IAX2, transcoding, NAT, or many other 
> things that make the two legs of the call not compatible with reinvites.
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