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?
On 5.1.2013 ?. 03:37 ?., 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).
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?
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 <boboodz at 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
Thanks What would you use to measure jitter / packetloss in real time?
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 <boboodz at 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
2013/1/5 joachim <zoachien at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130108/d58abaa9/attachment.htm>
Leandro Dardini
2013-Jan-08 09:40 UTC
[asterisk-users] Detect Low Quality Calls - Realtime
2013/1/8 Lenz Emilitri <lenz.loway at gmail.com>> > 2013/1/5 joachim <zoachien at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130108/4e647e60/attachment.htm>
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.loway at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Sent: Tuesday, January 8, 2013 2:25 PM Subject: Re: [asterisk-users] Detect Low Quality Calls - Realtime 2013/1/5 joachim <zoachien at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130108/e5764cb0/attachment.htm>