Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 22:32, Johan Wilfer skrev: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! I contacted them and will use a server connected to a switch-port in mirroring mode. The gui seems like a great tool in troubleshooting. Nobody uses the rtcp-stats in asterisk for quality monitoring? Other suggestions? -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
On Wed, Apr 11, 2012 at 4:29 AM, Johan Wilfer li...@jttech.se wrote: Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Cheap dedicated server with a span port on the switch. We *never* run anything other than Asterisk on a production voice server. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating different options. Aside from the manual thing I could see two variants: - Dump the traffic (on the server or another via switch port mirroring/monitoring) and analyze it with tshark - Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking for some ideas to setup this so I can eliminate this manual and time-consuming process in the future. And know about the problems before the customer complains about the quality.. Thanks in advance! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
Le 09/04/2012 13:42, Johan Wilfer a écrit : After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating different options. Aside from the manual thing I could see two variants: - Dump the traffic (on the server or another via switch port mirroring/monitoring) and analyze it with tshark - Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking for some ideas to setup this so I can eliminate this manual and time-consuming process in the future. And know about the problems before the customer complains about the quality.. At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. To monitor the traffic, you can use voipmonitor.org -- 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
Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.netwrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. To monitor the traffic, you can use voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
OpenVZ is not really virtualisation, though for some reason people insist on throwing it into the same discursive space as Xen, VMware, HyperV, etc. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Johan Wilfer li...@jttech.se wrote: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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
