Re: [Sipp-users] SIPp on Mandrake Linux

[<san...@wi...>]

Hi All,
=20
Following is the scenrio, I am planning to test.=20
System Under Test(SUT) is an advanced SIP Phone with Touch Sensitive =
Screen which is registered with Cisco Call Manager.
=20
1. The SIPp should originate the call to the SUT.
2. The call recieved by the SUT will be answered manually.
3. SIPp should send the audio stream towards the SUT and it should be =
heard at the SUT.
4. And also, when the person at the SUT end speaks, it should be =
captured by the SIPp.
=20
I think for the step 3 above, I can make use of "PCAP Play".=20
In this case, will the audio sent by SIPp, be heard at the SUT  as a =
normal voice message ? Pls. share with me your earlier experiences, tips =
and tricks...
=20
For Step 4 above, can I make use of "RTP Echo" and hence capture and =
play back the captured RTP stream to the SUT ?=20
=20
Pls. suggest.
=20
Thanks in advance.
=20
Regards,
Santosh
=20
=20
=20
=20

________________________________

From: S.K. Santoshkumar (WT01 - IP-Multimedia Carrier & Ent Networks)
Sent: Mon 7/31/2006 6:22 PM
To: sip...@li...
Subject: SIPp on Mandrake Linux


Hi,
I am trying to compile the SIPp on Mandrake linux:
The following link provides the pre-requisites for the compilation of =
SIPp and tips for each OS, but it doesn't talk abt Mandrake Linux.
http://sipp.sourceforge.net/wiki/index.php/Compilation
I am using Mandrake v10.0.
Has anybody tried on Mandrake Linux.  If yes, pls. provide the tips to =
use SIPp on Mandrake Linux.
Also, kindly let me know the source fm where I can get the following for =
Mandrake Linux ?

*	C++ compiler=20
*	Curses library=20
*	OpenSSL (>=3D0.9.7)=20

For pcap play (RTP) support:=20

*	libnet=20
*	libpcap=20

=20
Thanks,
Santosh
=20
________________________________

From: Makarand Bhagwat [mailto:mbh...@gm...]
Sent: Mon 7/24/2006 9:47 PM
To: S.K. Santoshkumar (WT01 - IP-Multimedia Carrier & Ent Networks)
Cc: sip...@li...
Subject: Re: [Sipp-users] Audio Support of SIPp


Answers in-line
Hope this helps

Regards,
Makarand


On 7/24/06, san...@wi... < san...@wi... =
<mailto:san...@wi...> > wrote:=20

	Hi All,
	=20
	The following link states that " SIPp can send audio and video RTP with =
any codec <http://forum/forum.php?forum_id=3D577392> " and I am =
interested in testing audio exchange between SIPp and a advanced SIP End =
Point.
	=20
	http://sourceforge.net/forum/forum.php?forum_id=3D577392=20


To run SIPP using audio you need to install the latest version of SIPP =
from=20
http://sipp.sourceforge.net/snapshots/=20

Compile SIPp using the instructions given @

http://sipp.sourceforge.net/doc1.1/reference.html#Installing+SIPp

Please note that you need to install Libnet for using SIPp with audio.=20
Also refer to documentation on how to use sipp with different scenarios. =
=20




	And in the note at the end of the article, it also mentions that this =
feature is available only on Linux systems.
	=20
	Does it mean that the System Under Test and the SIPp tool both should =
be running on Linux ? Or it just mentions that SIPp (to test audio =
media) has to be running on a linux machine and the System Under Test =
can be on any platform.


The system running SIPp should be running Linux. The System under Test =
could be running any OS.
=20


	Pls. clarify.. Also any information regarding testing the presence of =
audio when SIPp is in call with a SIP endpoint, will be highly =
appreciated.
	=20
	Thanks,
	Santosh

	=
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