[asterisk-users] Different box for SIP and RTP
Mohammad Khan
beeplove at gmail.com
Mon May 16 08:36:18 CDT 2011
Can't that third-party be an asterisk box?
After hand off RTP processing, does the first box (who, hand off) still in
charge of SIP packets?
On Mon, May 16, 2011 at 9:13 AM, Alex Balashov <abalashov at evaristesys.com>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.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> --
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