<br><br><div><span class="gmail_quote">On 11/16/07, <b class="gmail_sendername">Jim Dalton</b> <<a href="mailto:jim.dalton@transnexus.com">jim.dalton@transnexus.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;">
Each call has two legs: one call leg inbound to the B2BUA and one call leg<br>outbound from the B2BUA.<br><br>1500/400 calls<br>3000/800 call legs</blockquote><div><br><br>how was testing done to guarantee no audio distortions or anything else? I am going through the document but it appears that if a call was completed it was deemed good so far, and havent found anything relating to how calls can be done before audio starts to degrade. It is my belief that with media passing through the system, audio will degrade before you max out the systems ability to respond to SIP invites, as such how many calls a server can do with acceptable audio is more important than how many it can do in total since quality is important to customers, and poor quality will cost you many customers.
<br></div><br></div><br clear="all"><br>-- <br>Trixter <a href="http://www.0xdecafbad.com">http://www.0xdecafbad.com</a> Bret McDanel<br>Belfast +44 28 9099 6461 US +1 516 687 5200<br><a href="http://www.trxtel.com">
http://www.trxtel.com</a> the phone company that pays you!