[SIPForum-discussion] RES: Test of call scenarios withtheprotocol SIP

Adam Stein astein at mudynamics.com
Thu Oct 23 23:30:20 UTC 2008


Additional paid SIP testing details online at
http://www.mudynamics.com/solutions/next-gen-ip-services/sip.html

Regards.

Adam 

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Moshe Ostrovsky
Sent: Thursday, October 23, 2008 1:22 PM
To: Vicky; Giscard Fernandes Faria
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] RES: Test of call scenarios
withtheprotocol SIP

Hi.
 
I can suggest you ProLab, excellent tool, simple and user-friendly, very
good reporting, good support. 
 
I can organize it for you with a good price.
 
Tell me if you are interested.
 
Moshe.

________________________________

From: discussion-bounces at sipforum.org on behalf of Vicky
Sent: Thu 23/10/2008 07:59
To: Giscard Fernandes Faria
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] RES: Test of call scenarios with
theprotocol SIP


Please suggest some good paid tools for testing of both SIP UA and SIP
Proxy. 

-Vicky


On Mon, Oct 20, 2008 at 5:01 PM, Giscard Fernandes Faria
<GISCARDF at nec.com.br> wrote:


	Rajae,

	 

	IF you don't want to spend a bit of Money with a tool to test
SIP scenarions I would recommend you the SIPP.

	It is an easy to use, free and has some cool features like use
regex in order to identify fields.

	 

	Of course, it still lack some features to cover the whole SIP
protocol, but I believe you can use it for

	the most scenarios.

	 

	http://sipp.sourceforge.net/

	 

	Hope this can help.

	 

	Regards

	 

	
________________________________


	De: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] Em nome de rajae diki
	Enviada em: quinta-feira, 16 de outubro de 2008 16:57
	Para: discussion at sipforum.org
	Assunto: [SIPForum-discussion] Test of call scenarios with the
protocol SIP

	 

	Hi everybody,

	 

	I'm working on project about SIP with platform (Asterisk, Sipsak
and Sipfoundry), i have to test call scenarios between every entity in
SIP architecture with the use of the protocol SIP trunking.

	 

	If someone could help me and give me references.

	 

	Many thanks.

	 

	Rajae.

	 

	 


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