Asterisk is not a sip proxy but it *can* partly act as a sip proxy if reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk connects 2 end points directly and does signalling between them . <br>Asterisk is a PBX now suppose u need to record all calls ..do conferencing stuff then rtp stream need to pass from asterisk ( openser cant do this bcoz it just connects 2 endpoints and only does signalling ) .. If you do canreinvite=yes in
sip.conf for both peers then asterisk does only signalling ( also dial command should not have transfer parameters tT .. ) . If both peers are behind NAT then asterisk reinvites may not work properly .<br><br><div><span class="gmail_quote">
On 23/07/07, <b class="gmail_sendername">bilal ghayyad</b> <<a href="mailto:bilmar_gh@yahoo.com">bilmar_gh@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Dear Edgar;<br><br>I am little bit confused, do u mean that asterisk does<br>not work in that way:<br><br>RTP (media) to be from the sournce to the destination<br>directly while signaling to be via asterisk?<br><br>So, what he parameter canreinvite is doing?
<br><br>Regards,<br>------------<br>ITS<br>Ip Telephony and Contact Center Engineer<br>Eng. Bilal Ghayad<br>Mobile: 00965 9849460<br><br><br><br><br>Hello Asteriskers,<br><br>I'm confused about why Asterisk is not a SIP proxy and
<br>why exactly<br>this can affect the performance of a large Asterisk<br>system.<br><br>I know that Asterisk acts as a useragent endpoint, but<br>my doubt is why<br>exactly Asterisk could overload the call flow if the<br>
RTP voice stream<br>goes from the caller to the called party.<br><br>Does someone know how many calls or pencentaje that<br>could handle a SER<br>or OpenSER in comparison with Asterisk?<br><br><br><br><br><br><br>____________________________________________________________________________________
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