Can't that third-party be an asterisk box?<br>After hand off RTP processing, does the first box (who, hand off) still in charge of SIP packets?<br><br><br><div class="gmail_quote">On Mon, May 16, 2011 at 9:13 AM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div class="im">On 05/16/2011 09:00 AM, Mohammad Khan wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Is there way I can use two Asterisk box, one to maintain SIP packets and<br>
other for RTP traffic?<br>
</blockquote>
<br></div>
No, the signaling and bearer plane are integrated in Asterisk.<br>
<br>
But you can use reinvites to hand off RTP processing to third-party endpoints and bypass Asterisk, in qualifying call scenarios and network topologies.<br>
<br>
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</blockquote></div><br>