Re: [asterisk-users] VoIP sound quality : highroad sound
2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything. -- Johan Wilfer -- _ -- 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] VoIP sound quality : highroad sound
On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Kind regards, Jonas. -- _ -- 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] VoIP sound quality : highroad sound
I frequently use Audacity to analyze the audio data. In many cases I can see from the spectra (and other graphical representations) with what kind of problem I am dealing. Meanwhile, for most of my problems I no longer depend on an audio editor. I don't know whether this is helpful in your case. -- _ -- 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] VoIP sound quality : highroad sound
2013-11-13 11:55, Jonas Kellens skrev: On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Here is some suggestions to get started: http://www.enterprisenetworkingplanet.com/unified_communications/troubleshooting-common-sip-problems-with-wireshark.html Maybe one of your connections get congested? For example, if the two endpoints is your phone and the upstreams teleco. If the side from the teleco are bad and not the phone you need to take a closer look at the switches and routers on the way to the teleco. For example you can run tcpdump on your gateway to your ISP. If you see the problem here as well it may be your link or a upstreams problem. If you don't see it here it is somewhere in between.. Good luck! -- Johan Wilfer -- _ -- 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] VoIP sound quality : highroad sound
Hello, what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing. To be clear : the person calling is not standing next to a highway. But there seems to be a noise on the line of busy highroad that makes that the caller can not be understood. What can be causing this kind of poor quality ? Is it lack of resources on the Asterisk-server (codec translation ?) Is it lack of bandwith ? Is it a problem of CentOS (the underlying OS) ? Is it a physical problem of the server components (network interface ?) ? Kind regards, Jonas. -- _ -- 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] VoIP sound quality : highroad sound
On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 à 45 simultaneous calls (using G711a). Jonas. -- _ -- 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] VoIP sound quality : highroad sound
Did you have a look at the codecs that are involved? -- _ -- 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] VoIP sound quality : highroad sound
Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%) 0.0002 X.X.X.42 3c8956648ce 00:02:27 007301 00 ( 0.00%) 0. 007318 01 ( 0.01%) 0.0020 X.X.X.224 684333f5650 00:00:03 00 00 ( 0.00%) 0. 000178 00 ( 0.00%) 0. X.X.X.98 5eceb3a5624 00 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.9825ae26ee564 00:00:03 000179 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.986b26738a0c4 00:00:43 000137 00 ( 0.00%) 0. 000137 00 ( 0.00%) 0.0001 X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 X.X.X.1846893e957-fa 00:05:59 000576 00 ( 0.00%) 0. 000243 00 ( 0.00%) 0.0027 9 active SIP channels Thanks. Jonas. On 11/12/2013 05:32 PM, jg wrote: Are these all SIP-channels? If yes, or if one endpoint is always a SIP-device then you could issue a sip show channelstats in the cli. This is not exact, but it shows if you have any network or timing problems. I could say more about network problems, but first let's see what channelstats says. jg Am 12.11.2013 16:34, schrieb Jonas Kellens: On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 à 45 simultaneous calls (using G711a). Jonas. -- _ -- 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] VoIP sound quality : highroad sound
Yes, all SIP. Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%) 0.0002 X.X.X.42 3c8956648ce 00:02:27 007301 00 ( 0.00%) 0. 007318 01 ( 0.01%) 0.0020 X.X.X.224 684333f5650 00:00:03 00 00 ( 0.00%) 0. 000178 00 ( 0.00%) 0. X.X.X.98 5eceb3a5624 00 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.9825ae26ee564 00:00:03 000179 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.986b26738a0c4 00:00:43 000137 00 ( 0.00%) 0. 000137 00 ( 0.00%) 0.0001 X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 X.X.X.1846893e957-fa 00:05:59 000576 00 ( 0.00%) 0. 000243 00 ( 0.00%) 0.0027 9 active SIP channels Thanks. Jonas. On 11/12/2013 05:32 PM, jg wrote: Are these all SIP-channels? If yes, or if one endpoint is always a SIP-device then you could issue a sip show channelstats in the cli. This is not exact, but it shows if you have any network or timing problems. I could say more about network problems, but first let's see what channelstats says. jg Am 12.11.2013 16:34, schrieb Jonas Kellens: On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 à 45 simultaneous calls (using G711a). Jonas. -- _ -- 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
