Re: [asterisk-users] Sound quality issue
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 16 January 2011 12:28 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sound quality issue Le 15/01/2011 20:38, Cédric Lemarchand a écrit : 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. -- _ -- 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] Sound quality issue
Le 15/01/2011 20:38, Cédric Lemarchand a écrit : 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
[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. Cédric -- _ -- 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] Sound quality issue
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: Cédric Lemarchand cedric.lemarch...@ixcore.com To: asterisk-users@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. Cédric -- _ -- 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] Sound quality issue
Le 15/01/11 20:50, Sevana Oy a écrit : 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. Yes we already do some records. We don't have access to the internal network of the provider, so the network topology is quiet simple, only 2 sides : Asterisk site = MPLS NETWORK (the provider) = remote site Sound quality problems are present on both sent and received RTP flow. 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. Cédric -- _ -- 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] Sound quality issue
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 -- _ -- 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] Sound quality issue
Greetings everyone, I've been having some strange issues with my Asterisk box and some snom phones. In some cases, when I talk, the sound in the other end is cut off, I stop earing the background noise - looks like a walkie-talkie. I've tried this between phones in the same network and in all but one this happens. The one where it doesn't happen is the one connected directly to the router. I've tried different codecs with and without transcoding, including g722, and, on that phone, it all goes well. Could this have anything to do with the network? The main issue is that the router doesn't have QoS/ToS/whatever for me to test it. Plus, the phones are 3 switches away from the other phones. Thanks in advance. Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality issue in one direction and wctdm problem with APIC enabled kernel
On 2/28/06, Chris Miller [EMAIL PROTECTED] wrote: I'm chasing down a pop/click type of disturbance on a PBX system.Strangely, the disturbance is only heard by the outside caller, theinternal recipient hears the caller crystal clear. This seems to havecrept up when upgrading the zaptel driver to the 1.2 series whilerunning 1.0.10. I went ahead and upgraded the entire system to 1.2.4.That's funny, I'm trying to chase down pop/clicks on two of my installations but they are only heard by the internal recipient. The outside caller doesn't hear it at all! We have Dell SC420, Asterisk/Zaptel 1.2.1 and a TE110P integrated into a legacy PBX.It only occurs with internal users that are using their legacy handsets to call out through Asterisk. The Cisco 79XX users do not have this issue. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound quality issue in one direction and wctdm problem with APIC enabled kernel
I'm chasing down a pop/click type of disturbance on a PBX system. Strangely, the disturbance is only heard by the outside caller, the internal recipient hears the caller crystal clear. This seems to have crept up when upgrading the zaptel driver to the 1.2 series while running 1.0.10. I went ahead and upgraded the entire system to 1.2.4. The system is a ~2Ghz AMD 32bit system, with 512MB of memory and nothing other than Asterisk running. Phone traffic is minimal, perhaps 3 simultaneous calls max, but the problem occurs with just one call. It's located in a data center with ~20ms pings to the ITSP and ~20ms pings to the remote office IP phones. Up to this point, ztdummy was in use without problems, although the timing (zttest) was a hair under the recommended threshold. I dropped in a TDM400P for testing, and although the timing improved, the symptom remained. The system has an IDE drive, and I verified the hdparm dma/irq settings were enabled. The TDM card was sharing interrupts, so I recompiled the kernel with APIC support. Unfortunately the wctdm module will no longer load after recompile and install into the new kernel directory. I went back to the ztdummy driver with the same problem. Below is the relevant errors and info. Chris # modprobe wctdm FATAL: Error inserting wctdm (/lib/modules/2.6.12-prep/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm # dmesg wctdm: disagrees about version of symbol zt_receive wctdm: Unknown symbol zt_receive wctdm: disagrees about version of symbol zt_qevent_lock wctdm: Unknown symbol zt_qevent_lock wctdm: disagrees about version of symbol zt_ec_chunk wctdm: Unknown symbol zt_ec_chunk wctdm: disagrees about version of symbol zt_transmit wctdm: Unknown symbol zt_transmit wctdm: disagrees about version of symbol zt_unregister wctdm: Unknown symbol zt_unregister wctdm: disagrees about version of symbol zt_hooksig wctdm: Unknown symbol zt_hooksig wctdm: disagrees about version of symbol zt_register wctdm: Unknown symbol zt_register # cat /proc/interrupts CPU0 0: 34991774IO-APIC-edge timer 1: 10IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 12:111IO-APIC-edge i8042 14: 170392IO-APIC-edge ide0 15: 383872IO-APIC-edge ide1 18: 0 IO-APIC-level SiS SI7012, SiS SI7013 Modem 19: 164220 IO-APIC-level eth0 20: 0 IO-APIC-level ohci_hcd:usb2 21: 0 IO-APIC-level ohci_hcd:usb3 22: 0 IO-APIC-level ohci_hcd:usb4 23: 0 IO-APIC-level ehci_hcd:usb1 NMI: 0 LOC: 34991738 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
