Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check on his network equipments, everything is fine too, no packets loss recorded on routers's interfaces ect ... We have, on our side, check and replace all the VOIP equipments (spare rocks), an reduce the configuration to its simpliest (MPLS router <=> ethernet cable <=> VOIP equipment), quality problem still there. I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? Any help would be greatly appreciated, thx. C?dric
Hello, Can you record audio at different locations on its route? Our experience would suggest (of course) using intrusive or non-intrusive perceptual voice quality evaluation at different parts of the network to localize the one where it drops down. Best regards, Sevana Oy http://www.sevana.fi http://twitter.com/sevana ----- Original Message ----- From: "C?dric Lemarchand" <cedric.lemarchand at ixcore.com> To: <asterisk-users at lists.digium.com> Sent: Saturday, January 15, 2011 10:38 PM Subject: [asterisk-users] Sound quality issue> Hello, > > Our Asterisk runs with multiple remote sites (12 over an MPLS network), > everything works fine except for the last site we have juste installed. > > When VOIP flows comes/goes from/to this site, there are sound quality > issues, persistent, 100% reproducible, on every call. This is not a > bandwidth or latency or jitter problem, everything is fine on the network. > Our MPLS provider does all check on his network equipments, everything > is fine too, no packets loss recorded on routers's interfaces ect ... > We have, on our side, check and replace all the VOIP equipments (spare > rocks), an reduce the configuration to its simpliest (MPLS router <=> > ethernet cable <=> VOIP equipment), quality problem still there. > > I am sure there are RTP packets losses somewhere, except RTP debug in > the asterisk CLI, how can i determine where the problem come from ? > > Any help would be greatly appreciated, thx. > > C?dric > > -- > _____________________________________________________________________ > -- 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
> I am sure there are RTP packets losses somewhere, except RTP debug in > the asterisk CLI, how can i determine where the problem come from ?If it is possible to make a network trace in a Wireshark compatible format, Wireshark can parse all the SIP and RTP messaging and give you lots of statistics, including packet loss, jitter, etc. Check the Wireshark site (http://www.wireshark.org/) for more information. -- Andreas Sikkema
Le 15/01/2011 20:38, C?dric Lemarchand a ?crit :> Hello, >Hi> [...] > I am sure there are RTP packets losses somewhere, except RTP debug in > the asterisk CLI, how can i determine where the problem come from ? >[...] You don't tell which protocol (SIP, IAX, H323) nor which asterisk version. FYI Asterisk 1.6.2.15 in iax had audio quality problems, solved in 1.6.2.16. If you have the possibility, connect directly a phone to the server, eg Device - LAN (no MPLS) - Asterisk. Check also if a call to echo test has the same bad quality. -- Daniel
Something that often gets forgotten is the on-site LAN infrastructure as well. It could be a bad/faulty switch, rubbish cabling, induced interference etc. etc. all at the customers premises. Maybe a handset plugged directly in to the back of the router, before it hits the LAN would tell you whether the call is actually getting 'distorted' en-route or not? -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 16 January 2011 12:28 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Sound quality issue Le 15/01/2011 20:38, C?dric Lemarchand a ?crit :> Hello, >Hi> [...] > I am sure there are RTP packets losses somewhere, except RTP debug in > the asterisk CLI, how can i determine where the problem come from ? >[...] You don't tell which protocol (SIP, IAX, H323) nor which asterisk version. FYI Asterisk 1.6.2.15 in iax had audio quality problems, solved in 1.6.2.16. If you have the possibility, connect directly a phone to the server, eg Device - LAN (no MPLS) - Asterisk. Check also if a call to echo test has the same bad quality. -- Daniel -- _____________________________________________________________________ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085.