[SIPForum-discussion] SIPp Client, OPENSIPS and SIPp server on the same machine.

Tue Mar 9 04:41:08 UTC 2010

Hi,

We have used SIPp UAC and UAS in linux environment. Execution wise,
there could be some differences. SIPP UAC can talk to OPenSIPS and in
turn OpenSIPS should have outbound proxy referring to SIPP UAS. I am
here with listing one example how SIPP UAC and UAS can be used. 

client: 

./sipp 127.0.0.1:5060 -p 6001 -d 10000 -t u1 -mp 30101 -f 5 -mi
127.0.0.1 -sn 'branchc'  

-p Signifies local port

-mp media port

-mi media ip

server: 

./sipp 127.0.0.1:5061 -p 5063 -s 8880002 -t u1 -mp 30102 -mi 127.0.0.1
-sn 'branchs'

-p Signifies local port

-mp media port

-mi media ip 

However since we are using in loop back mode no remote port and ip is
necessary.

However if hosted on different PC we need -rsa host:port parameter

e.g. hosted by 21.21.21.10:5061

Server:

./sipp 21.21.21.10:5061 -p 5061 -s 8880002 -t u1 -mp 30102 -mi
21.21.21.10 -sn 'branchs'

client: 

./sipp 172.16.52.92:5060 -p 6001 -rsa 21.21.21.10:5061 -t u1 -mp 30101
-f 5 -mi 172.16.52.92 -sn 'branchc'

Similar options could be available in SIPP windows version. 

Rgds,

Narasimham.

________________________________

From: Udayarajan G [mailto:uday_software_testing at yahoo.co.in] 
Sent: Monday, March 08, 2010 2:30 PM
To: discussion at sipforum.org; Narasimham Settipalli (WT01 - Telecom
Equipment)
Subject: RE: [SIPForum-discussion] SIPp Client,OPENSIPS and SIPp server
on the same machine.

Hi Narasimham Settipali 

              The proxy is OPENSIPS and not SIPp Proxy. Can you let me
know

How to configure the elements on different UDP ports.and what
configuration is needed on 

the XML file. It would be very useful as I am new to SIP.

By the way I am a X-Wiproite having worked in Wipro for around 6-7
years.

Best Regards

Uday

--- On Sun, 7/3/10, narasimham.settipalli at wipro.com
<narasimham.settipalli at wipro.com> wrote:

	From: narasimham.settipalli at wipro.com
<narasimham.settipalli at wipro.com>
	Subject: RE: [SIPForum-discussion] Invite not reaching the IAD
or SIP ATA.
	To: uday_software_testing at yahoo.co.in, discussion at sipforum.org
	Date: Sunday, 7 March, 2010, 12:56 PM

	HI,

	Yes. It is possible to run SIPp UAC, SIP Proxy and SIPp UAS in
same machine provided they use different UDP ports. 

	SIPP has command line options where we can mention next level IP
address and port to be contacted. It may be need to configure XML files
provided by SIPp. 

	Rgds,

	Narasimham.

________________________________

	From: discussion-bounces at sipforum.org on behalf of Udayarajan G
	Sent: Sat 3/6/2010 5:53 PM
	To: discussion at sipforum.org
	Subject: Re: [SIPForum-discussion] Invite not reaching the IAD
or SIP ATA.

Hi All

I have installed SIPp on my machine. I have installed Asterix on my
machine.

The machine runs on Windows XP.Is it possible to replicate the bellow
scenario from 

a single machine using OPENSER or ASTERIX for learning/testing purpose.

<SIPp UAC>-----<OPENSER(Proxy)>--------<SIPp UAS> 

Regards

Uday

--- On Wed, 3/3/10, Vijay Tiwari <vijay11tiwari at gmail.com> wrote:

	From: Vijay Tiwari <vijay11tiwari at gmail.com>
	Subject: Re: [SIPForum-discussion] Invite not reaching the IAD
or SIP ATA.
	To: "Juan Carlos Garcia" <juan.garcia.aro at gmail.com>
	Cc: discussion at sipforum.org
	Date: Wednesday, 3 March, 2010, 10:39 AM

	Hello JC

	Same kind of issue we are facing. and we are come out of this
issue by checking same DID on other IP address. and we found the problem
between rouiting issue will two ISP here. 

	and one more thing i want to share with you that. we have two
switches one in india and other one in US. when i am sending packet to
india switch to gateway no packet recieved. but at the same time when i
send packet from US switch in working perfectly ok. 

	so we come to the point that  there is some routing problem in
inter ISP.

	Thanks

	vijay

	On Tue, Mar 2, 2010 at 9:42 PM, Juan Carlos Garcia
<juan.garcia.aro at gmail.com
<http://in.mc946.mail.yahoo.com/mc/compose?to=juan.garcia.aro@gmail.com>
> wrote:

	Hello Everyone;

	I am having a SIP terminal were the inbound calls are not coming
through. If i enabled a wireshar trace in the DSLAM for instance i can
see the invite going out of the softswitch ( a lot of invites messages )
but i see nothing coming to my SIP terminal i enabled sip debug in my
terminal but there is no ip packet coming or being dropped?

	It could be a routing issue between my SIP Agent and the
softswitch???

	Thanks for any info..

	JC

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