[asterisk-users] Different box for SIP and RTP

[Leif Madsen]

On 11-05-16 09:13 AM, Alex Balashov wrote:
> On 05/16/2011 09:00 AM, Mohammad Khan wrote:
> 
>> Is there way I can use two Asterisk box, one to maintain SIP packets and
>> other for RTP traffic?
> 
> No, the signaling and bearer plane are integrated in Asterisk.
> 
> But you can use reinvites to hand off RTP processing to third-party endpoints
> and bypass Asterisk, in qualifying call scenarios and network topologies.

You could try directrtpsetup=yes which is similar to directmedia, except the
audio is redirected in the initial INVITEs rather than reinviting the media a
few RTP packets in.

Leif.