On Aug 2 2007, John Meksavan wrote: >Asterisk Users, > > I recently ran into some problems with the quality of service with > Teliax. > This occurred on August 1, 2007 with a dropped outbound call, audio > quality isse on the callee side- not hearing me well on callee side, and > sending DTMF tones (configured for RFC2833). Am I the only Teliax > customer having this problem? > > It seems like when I am ready to go live with my Asterisk PBX System, I >run into quality of service issues with the SIP provider. Who should I go >with that would guarantee me quality service just like an analog line?
VoIP is susceptible to packet delivery problems anywhere between your PBX and your SIP provider's PRI lines/termination point. If you have direct SIP PBX to SIP PBX calls, then your problems can be anywhere on the Internet path between the sites. The only workaround that I know of is having your ISP be your SIP provider, so that your SIP packets only cross your ISP's own network to its termination point, and do not cross the public Internet. This way QoS can work from your office to your ISP's office to make sure you maintain reliability. I have not personally used iTEL-ip's 'iTEL Voice Service', but others have said, as do their own notes that their network QoS is effective at maintaining call quality. When I contacted them, their pricing for a 'QoS private IP backbone for voice and data' was $618/month for a full 1.5mbps T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending on contract length. Per minute rates were $.03. When I ran the numbers, it appeared that a regular full T1 + a regular full PRI would be only slightly more. A major tradeoff comes in the physical location flexibility you get with SIP over traditional phone lines in the case you need to move an office (although physically moving the phones to a non iTEL-ip data line would mean you're not getting their Qos). iTEL-ip's 'iTEL Voice Service' http://www.itelconnect.com/default.aspx?type=t§ion=iTEL-ipVoiceService&selection=16 http://wiki.pbxnsip.com/index.php/ITEL-ip -hk _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
