Suggestion: learn to use the facility in Wireshark that can log a SIP/RTP stream and report actual latency and packet delivery stats. That will give you some solid info on at least one aspect of call quality.
Message: 2 Date: Tue, 15 Dec 2009 11:03:07 -1000 From: "Ben Schorr" <b...@rolandschorr.com> Subject: Re: [asterisk-users] Can't get G.729 to work... To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <02bd9be03009b24b9bf9e00257ad8103012d7...@hnl-pai-exh001.pacificatelier.com> Content-Type: text/plain; charset="US-ASCII"Yes, the routers are another issue we're dealing with. We've configured them to prioritize traffic to/from our Asterisk server but I'm not convinced that setting is really working as expected. So we're working with the vendor on that. The effective bandwidth is 6Mb/sec down x 1.4Mb/sec up (so basically 1.4x1.4 on the VPN). For 8 users, where rarely more than 2-3 of them are on the phone at any given time, that should be sufficient I think. They DO have to share the connection with their web browsing and e-mail and such but as best we've been able to tell they aren't saturating their connections - usually not more than 4-5 of the 8 are using their computers at any given time and most of them just do e-mail and local apps that shouldn't touch the Internet connection. Frankly I'm puzzled that they have these issues and the problems rarely seem to happen when I call them. I'll go to their site and make a few calls from one of their phones and it sounds perfect to me. But three days later all I hear is how frustrated they are because these new VOIP phones suck and they can "never" hear anybody and... <sigh>
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