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[asterisk-users] SiP call generator


 
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kchehab at xplorium.com
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PostPosted: Mon Feb 18, 2008 5:24 am    Post subject: [asterisk-users] SiP call generator Reply with 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 .





Regards




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atis at iq-labs.net
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PostPosted: Mon Feb 18, 2008 7:11 am    Post subject: [asterisk-users] SiP call generator Reply with 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

--
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





PostPosted: Tue Feb 19, 2008 2:04 am    Post subject: [asterisk-users] SiP call generator Reply with 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.

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





PostPosted: Tue Feb 19, 2008 10:00 am    Post subject: [asterisk-users] SiP call generator Reply with 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:
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--
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





PostPosted: Tue Feb 19, 2008 2:57 pm    Post subject: [asterisk-users] SiP call generator Reply with 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
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atis at iq-labs.net
Guest





PostPosted: Wed Feb 20, 2008 6:54 am    Post subject: [asterisk-users] SiP call generator Reply with quote

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





PostPosted: Wed Feb 20, 2008 1:35 pm    Post subject: [asterisk-users] SiP call generator Reply with 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?
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.

Quote:
Quote:
Quote:
Atis

Quote:
Quote:
Alex Balashov

--
Alex Balashov
--

(C) Matthew Rubenstein
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tzafrir.cohen at xorco...
Guest





PostPosted: Wed Feb 20, 2008 4:51 pm    Post subject: [asterisk-users] SiP call generator Reply with 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.

--
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





PostPosted: Wed Feb 20, 2008 4:51 pm    Post subject: [asterisk-users] SiP call generator Reply with quote

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





PostPosted: Wed Feb 20, 2008 5:56 pm    Post subject: [asterisk-users] SiP call generator Reply with 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???

--
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






PostPosted: Wed Feb 20, 2008 7:58 pm    Post subject: [asterisk-users] SiP call generator Reply with quote

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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