Hello, Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? Thanks, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110516/ab82dcfe/attachment.htm>
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/
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.