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<DIV dir=ltr align=left><SPAN class=484545215-02032007><FONT face=Arial
color=#0000ff size=2>I think he meant bundling a group of sip calls destine to
the same server or proxy so all the rtp headers can be eliminated except one.
However, IAX does that already.</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] <B>On Behalf Of </B>Jonson
Player<BR><B>Sent:</B> Friday, March 02, 2007 3:14 AM<BR><B>To:</B> Asterisk
Developers Mailing List<BR><B>Subject:</B> Re: [asterisk-dev] SIP
trunking<BR></FONT><BR></DIV>
<DIV></DIV>What you mean SIP Trunking?<BR><BR>
<DIV><SPAN class=gmail_quote>On 3/1/07, <B class=gmail_sendername>Anton</B>
<<A href="mailto:anton.vazir@gmail.com">anton.vazir@gmail.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Guys,<BR><BR>any
plans to support SIP
Trunking?<BR>_______________________________________________<BR>--Bandwidth
and Colocation provided by <A href="http://Easynews.com">Easynews.com</A>
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