[asterisk-users] Fw: Stress testing Asterisk
Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? - Forwarded Message - From: Marie Fischer ma...@vtl.ee To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress testing Asterisk On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote: Hi, I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s extension_to_dial option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- 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
[asterisk-users] Fw: Stress testing Asterisk
From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Mitul On Wednesday, May 22, 2013, Tommy Cooper wrote: Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? - Forwarded Message - From: Marie Fischer ma...@vtl.ee To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress testing Asterisk On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote: Hi, I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s extension_to_dial option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- 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 -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422-- _ -- 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] Fw: Stress testing Asterisk
I believe there are options for rtp / audio.. Look at pcap play and rtp echo... Transcoding would be another beast - if you are allowing it Sent from my iPhone 5 On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Mitul On Wednesday, May 22, 2013, Tommy Cooper wrote: Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? - Forwarded Message - From: Marie Fischer ma...@vtl.ee To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress testing Asterisk On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote: Hi, I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s extension_to_dial option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- 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 -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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] Fw: Stress testing Asterisk
On 13-05-22 10:02 AM, Tommy Cooper wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Once upon a time, we set out to create exactly this for testing asterisk. Our goal would have been to run the test every week, comparing the results from the previous week, to make sure asterisk's performance was not getting worse as new commits happened. We came up with the idea of loading testing asterisk using SIPp or some other dialer, then determining at what point asterisk would start failing (performance). We decided the point of failure was quality of audio, since it is usually the first thing to go (even though call control still works). It took a while, but with the help of Leif, we found a tool to analyse audio streams (using MOS score[1]). Basically, you take the original audio file, play it across the network, then record the other side. Then, comparing the two files via Aqua, you get your MOS score. If the score was less then x, you knew asterisk was hitting a performance limit. Track that over time and concurrent calls, you have your metrics. [1] http://www.sevana.fi/aqua.php -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Fw: Stress testing Asterisk
El 22/05/13 12:25, Paul Belanger escribió: On 13-05-22 10:02 AM, Tommy Cooper wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Once upon a time, we set out to create exactly this for testing asterisk. Our goal would have been to run the test every week, comparing the results from the previous week, to make sure asterisk's performance was not getting worse as new commits happened. We came up with the idea of loading testing asterisk using SIPp or some other dialer, then determining at what point asterisk would start failing (performance). We decided the point of failure was quality of audio, since it is usually the first thing to go (even though call control still works). It took a while, but with the help of Leif, we found a tool to analyse audio streams (using MOS score[1]). Basically, you take the original audio file, play it across the network, then record the other side. Then, comparing the two files via Aqua, you get your MOS score. If the score was less then x, you knew asterisk was hitting a performance limit. Track that over time and concurrent calls, you have your metrics. [1] http://www.sevana.fi/aqua.php Hi! I haven't used it, but there is a quality test algorithm provided by ITU. http://stackoverflow.com/questions/2329403/how-to-start-a-voice-quality-pesq-test http://en.wikipedia.org/wiki/PESQ http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=arnumber=6043771queryText%3DDevelopment+of+a+Speech+Quality+Monitoring+Tool+based+on+ITU-T+P.862 - CeSPI Centro Superior para el Procesamiento de la Información Universidad Nacional de La Plata --- Proteja el Medioambiente. No imprima este mail si no es absolutamente necesario -- _ -- 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
