<div>I think you can use asterisk for your sip implementation.</div>
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<div>As you already know, SIP is peer to peer and you don't need asterisk when u want to communicate just between two users. If you are thinking about having more than 2 users (say 25) in the setup, Asterisk will help you there. In this case, Asterisk will be the SIP registrar/feature server for all the users. Sip UA (users) will send the requests to asterisk and asterisk will route the call appropriately.
<br><br>Hope this helps</div>
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<div><span class="gmail_quote">On 11/3/06, <b class="gmail_sendername">Christopher Chan</b> <<a href="mailto:chrisfz@netvigator.com">chrisfz@netvigator.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>> so if you have u1 and u2 and they simply wish to<br>> communicate and they have eachother's IP, do they need
<br>> Asterisk for something?<br><br>no.<br><br>><br>> Also, in a pure SIP implementation, what does Asterisk<br>> offer and what is Asterisk required for?<br><br>line/no. registration and setting up calls between sip phones.
<br><br>asterisk is required, I believe, when you need an easy method of getting<br>your users to be able to call one another other than supplying a list of<br>ip addresses and fixing the ip address of each user.<br>_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Thanks,<br>Vikki.