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kchehab at xplorium.com Guest
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Posted: Mon Feb 18, 2008 5:24 am Post subject: [asterisk-users] SiP call generator |
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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
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atis at iq-labs.net Guest
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Posted: Mon Feb 18, 2008 7:11 am Post subject: [asterisk-users] SiP call generator |
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On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
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abalashov at evaristes... Guest
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Posted: Tue Feb 19, 2008 2:04 am Post subject: [asterisk-users] SiP call generator |
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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:
Quote: | On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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 |
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atis at iq-labs.net Guest
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Posted: Tue Feb 19, 2008 10:00 am Post subject: [asterisk-users] SiP call generator |
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On 2/19/08, Alex Balashov <abalashov at evaristesys.com> wrote:
Quote: | 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
Quote: |
Atis Lezdins wrote:
Quote: | On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
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abalashov at evaristes... Guest
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Posted: Tue Feb 19, 2008 2:57 pm Post subject: [asterisk-users] SiP call generator |
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Just out of curiosity, why PHP?
Atis Lezdins wrote:
Quote: | On 2/19/08, Alex Balashov <abalashov at evaristesys.com> wrote:
Quote: | 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
Quote: | Atis Lezdins wrote:
Quote: | On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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 |
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atis at iq-labs.net Guest
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Posted: Wed Feb 20, 2008 6:54 am Post subject: [asterisk-users] SiP call generator |
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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 <abalashov at evaristesys.com> wrote:
Quote: | Just out of curiosity, why PHP?
Atis Lezdins wrote:
Quote: | On 2/19/08, Alex Balashov <abalashov at evaristesys.com> wrote:
Quote: | 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
Quote: | Atis Lezdins wrote:
Quote: | On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
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email at mattruby.com Guest
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Posted: Wed Feb 20, 2008 1:35 pm Post subject: [asterisk-users] SiP call generator |
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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:
Quote: | On 2/19/08, Alex Balashov <abalashov at evaristesys.com> wrote:
Quote: | 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.
|
Quote: | Atis
Quote: | Atis Lezdins wrote:
Quote: | On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote:
Quote: |
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.
|
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--
Alex Balashov
--
(C) Matthew Rubenstein |
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tzafrir.cohen at xorco... Guest
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Posted: Wed Feb 20, 2008 4:51 pm Post subject: [asterisk-users] SiP call generator |
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On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote:
Quote: | 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:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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atis at iq-labs.net Guest
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Posted: Wed Feb 20, 2008 4:51 pm Post subject: [asterisk-users] SiP call generator |
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On 2/20/08, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote:
Quote: | 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.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
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tzafrir.cohen at xorco... Guest
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Posted: Wed Feb 20, 2008 5:56 pm Post subject: [asterisk-users] SiP call generator |
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On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote:
Quote: | Quote: | Test of audio quality is something I'm not really sure how to do.
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Run tests, and ChanSpy() them? See at which point decrease of quality
becomes hearable.
|
Manually???
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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Guest
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Posted: Wed Feb 20, 2008 7:58 pm Post subject: [asterisk-users] SiP call generator |
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Sure, run 10 concurrently and see how it sounds. Scale up by a factor
of 10 until it sounds crappy then start scaling down. <shrug> At least
I think that's what Atis meant.
Moj
Tzafrir Cohen wrote:
Quote: | On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote:
Quote: | Quote: | 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???
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