Re: [asterisk-users] SiP call generator
Sure, run 10 concurrently and see how it sounds. Scale up by a factor of 10 until it sounds crappy then start scaling down.At least I think that's what Atis meant. Moj Tzafrir Cohen wrote: > On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote: > > >>> Test of audio quality is something I'm not really sure how to do. >>> >> Run tests, and ChanSpy() them? See at which point decrease of quality >> becomes hearable. >> > > Manually??? > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote: > > Test of audio quality is something I'm not really sure how to do. > > Run tests, and ChanSpy() them? See at which point decrease of quality > becomes hearable. Manually??? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
On 2/20/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote: > > Is there a simple tool that I can use to script Asterisk generating > > lots of calls according to a peak traffic curve, with random variance > > within a specified percentage around that curve, to test a number of > > DIDs at which I terminate voice recordings to test the audio and call > > quality? Any that will also give me a report of the actual traffic > > connections? > > > Most of the things here are probably not that difficult to script within > Asterisk itself, or with a simple wrapper. > > Test of audio quality is something I'm not really sure how to do. Run tests, and ChanSpy() them? See at which point decrease of quality becomes hearable. Regards, Atis -- Atis Lezdins VoIP Project Manager, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote: > Is there a simple tool that I can use to script Asterisk generating > lots of calls according to a peak traffic curve, with random variance > within a specified percentage around that curve, to test a number of > DIDs at which I terminate voice recordings to test the audio and call > quality? Any that will also give me a report of the actual traffic > connections? Most of the things here are probably not that difficult to script within Asterisk itself, or with a simple wrapper. Test of audio quality is something I'm not really sure how to do. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
Is there a simple tool that I can use to script Asterisk generating lots of calls according to a peak traffic curve, with random variance within a specified percentage around that curve, to test a number of DIDs at which I terminate voice recordings to test the audio and call quality? Any that will also give me a report of the actual traffic connections? On Tue Feb 19 09:00:45 CST 2008 Atis Lezdins wrote: > On 2/19/08, Alex Balashov wrote: >> Or, you can write your own scripts to generate calls via the Manager >> API, or use Asterisk call files (see voip-info.org on this topic). >> >> But, all other things being equal, it is probably preferred to use some >> sort of testing framework of the sort mentioned below. > > The PBX Testing Framework i mentioned (and also developed) provides > call-generation trough call-files so all you have to do is code action > scripts (answer, talk for 3-10 minutes, transfer to other extension, > etc..) and call generation scripts (random agent call every 10-20 > seconds, and random customer call every 20-30 seconds), all in PHP > with some functions and objects to make interaction easy. > Atis > >> Atis Lezdins wrote: >>> On 2/18/08, Khaled Chehab wrote: I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . >>> If you want just simple calls, i suppose SIPP can do that. >>> http://sipp.sourceforge.net/ >>> >>> If you want to have those calls perform some actions (send DTMF, etc), >>> you can try to write your own scripts based on PBX Testing Framework. >>> http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's >>> designed for testing queue-agents scenarios but i'm sure you can >>> adapt. >>> Atis >> Alex Balashov -- Alex Balashov -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
Well, PHP is language in which i'm coding most for last 5 years, so when i needed something fast, i took it. And maybe some day it will have web interface. Regards, Atis On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote: > Just out of curiosity, why PHP? > > Atis Lezdins wrote: > > On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote: > >> Or, you can write your own scripts to generate calls via the Manager > >> API, or use Asterisk call files (see voip-info.org on this topic). > >> > >> But, all other things being equal, it is probably preferred to use some > >> sort of testing framework of the sort mentioned below. > > > > The PBX Testing Framework i mentioned (and also developed) provides > > call-generation trough call-files so all you have to do is code action > > scripts (answer, talk for 3-10 minutes, transfer to other extension, > > etc..) and call generation scripts (random agent call every 10-20 > > seconds, and random customer call every 20-30 seconds), all in PHP > > with some functions and objects to make interaction easy. > > > > Regards, > > Atis > > > >> Atis Lezdins wrote: > >>> On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote: > > > I want to have a PC-based real-time VoIP bulk call generator (including > both > SIP signaling and RTP generation) > > for stress testing and precise analysis of the VoIP network equipment. > > > > Do any one knows a free program can do that . > >>> If you want just simple calls, i suppose SIPP can do that. > >>> http://sipp.sourceforge.net/ > >>> > >>> If you want to have those calls perform some actions (send DTMF, etc), > >>> you can try to write your own scripts based on PBX Testing Framework. > >>> http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's > >>> designed for testing queue-agents scenarios but i'm sure you can > >>> adapt. > >>> > >>> Regards, > >>> Atis > >>> > >> > >> -- > >> Alex Balashov > >> Evariste Systems > >> Web: http://www.evaristesys.com/ > >> Tel: (+1) (678) 954-0670 > >> Direct : (+1) (678) 954-0671 > >> Mobile : (+1) (706) 338-8599 > >> > >> ___ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
Just out of curiosity, why PHP? Atis Lezdins wrote: > On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote: >> Or, you can write your own scripts to generate calls via the Manager >> API, or use Asterisk call files (see voip-info.org on this topic). >> >> But, all other things being equal, it is probably preferred to use some >> sort of testing framework of the sort mentioned below. > > The PBX Testing Framework i mentioned (and also developed) provides > call-generation trough call-files so all you have to do is code action > scripts (answer, talk for 3-10 minutes, transfer to other extension, > etc..) and call generation scripts (random agent call every 10-20 > seconds, and random customer call every 20-30 seconds), all in PHP > with some functions and objects to make interaction easy. > > Regards, > Atis > >> Atis Lezdins wrote: >>> On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote: I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . >>> If you want just simple calls, i suppose SIPP can do that. >>> http://sipp.sourceforge.net/ >>> >>> If you want to have those calls perform some actions (send DTMF, etc), >>> you can try to write your own scripts based on PBX Testing Framework. >>> http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's >>> designed for testing queue-agents scenarios but i'm sure you can >>> adapt. >>> >>> Regards, >>> Atis >>> >> >> -- >> Alex Balashov >> Evariste Systems >> Web: http://www.evaristesys.com/ >> Tel: (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote: > Or, you can write your own scripts to generate calls via the Manager > API, or use Asterisk call files (see voip-info.org on this topic). > > But, all other things being equal, it is probably preferred to use some > sort of testing framework of the sort mentioned below. The PBX Testing Framework i mentioned (and also developed) provides call-generation trough call-files so all you have to do is code action scripts (answer, talk for 3-10 minutes, transfer to other extension, etc..) and call generation scripts (random agent call every 10-20 seconds, and random customer call every 20-30 seconds), all in PHP with some functions and objects to make interaction easy. Regards, Atis > > Atis Lezdins wrote: > > On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote: > >> > >> > >> > >> I want to have a PC-based real-time VoIP bulk call generator (including > >> both > >> SIP signaling and RTP generation) > >> > >> for stress testing and precise analysis of the VoIP network equipment. > >> > >> > >> > >> Do any one knows a free program can do that . > > > > If you want just simple calls, i suppose SIPP can do that. > > http://sipp.sourceforge.net/ > > > > If you want to have those calls perform some actions (send DTMF, etc), > > you can try to write your own scripts based on PBX Testing Framework. > > http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's > > designed for testing queue-agents scenarios but i'm sure you can > > adapt. > > > > Regards, > > Atis > > > > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
Or, you can write your own scripts to generate calls via the Manager API, or use Asterisk call files (see voip-info.org on this topic). But, all other things being equal, it is probably preferred to use some sort of testing framework of the sort mentioned below. Atis Lezdins wrote: > On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote: >> >> >> >> I want to have a PC-based real-time VoIP bulk call generator (including both >> SIP signaling and RTP generation) >> >> for stress testing and precise analysis of the VoIP network equipment. >> >> >> >> Do any one knows a free program can do that . > > If you want just simple calls, i suppose SIPP can do that. > http://sipp.sourceforge.net/ > > If you want to have those calls perform some actions (send DTMF, etc), > you can try to write your own scripts based on PBX Testing Framework. > http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's > designed for testing queue-agents scenarios but i'm sure you can > adapt. > > Regards, > Atis > -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote: > > > > > I want to have a PC-based real-time VoIP bulk call generator (including both > SIP signaling and RTP generation) > > for stress testing and precise analysis of the VoIP network equipment. > > > > Do any one knows a free program can do that . If you want just simple calls, i suppose SIPP can do that. http://sipp.sourceforge.net/ If you want to have those calls perform some actions (send DTMF, etc), you can try to write your own scripts based on PBX Testing Framework. http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's designed for testing queue-agents scenarios but i'm sure you can adapt. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SiP call generator
I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
