Re: [asterisk-users] Detect Low Quality Calls - Realtime
2013/1/5 joachim zoach...@securax.org You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
2013/1/8 Lenz Emilitri lenz.lo...@gmail.com 2013/1/5 joachim zoach...@securax.org You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. For MOS calculation I use voipmonitor, but it computer it at the end of the call. The voipmonitor guy is very handsome, maybe you can sponsor a patch to have the MOS calculation in real time. An external software can get it and halt the call if needed. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. -- You need a lot of data to calculate a MOS score, you will need the actual call. The only solution i can think of is that the phones start a fake call as soon as they are in focus and the server calculates some scores based on the fake call. When the client calls, the fake call is terminated and replaced with a real call. About the qualify, i don't know how to get the timing results from within the dialplan, i'm not even sure it's possible without patching. Z. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
When i worked in an internet provider with asterisk telephony solution - we used Aqua (http://www.sevana.fi) to measure voice quality. several nettops were spread across our network. The nettop called to our asterisk, the asterisk saved this voice file to the disk, then this file was sent to a server with Aqua software which compared this file to its original. then the quality (measured in percents) were sent to Zabbix monitoring. actually this data was used for analisys and it compares two files (not realtime). BR, Dmitry Pavlenko From: Lenz Emilitri lenz.lo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 8, 2013 2:25 PM Subject: Re: [asterisk-users] Detect Low Quality Calls - Realtime 2013/1/5 joachim zoach...@securax.org You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Thanks What would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Sometimes just the act of collecting performance data degrades the quality Sent from my iPhone 5 On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote: Thanks What would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
On 5.1.2013 г. 03:37 ч., XBrian wrote: I can only detect calls as they hit our server, do the magic and based on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call should be let through. I will accept all MOS values of 4.0 You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Joachim, thanks for the reply - delay you can somewhat estimate prior to the call (with qualify for example) Pls be explicit. How do I use qualify to measure delay - The jitter / packetloss you can only figure out when the call is already up for a while. what would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Asterisk sip show peers lists the qualify value in ms (milliseconds). Please read up on this and the setting for it in sip.conf config file Sent from my iPhone 5 On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote: Joachim, thanks for the reply - delay you can somewhat estimate prior to the call (with qualify for example) Pls be explicit. How do I use qualify to measure delay - The jitter / packetloss you can only figure out when the call is already up for a while. what would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detect Low Quality Calls - Realtime
Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a challenging and tricky one as I cannot install any software on the callers endpoint. I can only detect calls as they hit our server, do the magic and based on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call should be let through. I will accept all MOS values of 4.0 Any bright ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
