That i have done but it is giving following error:
/ats/tools/sipp-2.0.1.src> ./sipp 10.34.20.78 -sf
/home//CQ78827_UPDATE_pcap2.xml -m 1 -i 10.128.254.81 -p 6060 -mp 10000
In pcap pcap/g711a.pcap, npkts 236
max pkt length 260
base port 2006
In pcap pcap/dtmf_2833_1.pcap, npkts 10
max pkt length 24
base port 10000
On 12/08/2008, Srivastava, Anuj Kumar <[EMAIL PROTECTED]> wrote:
>
> Sipp should be compiled with pcap option.
> there's no specific command for executing pcap scenarios.
>
> regards
> Anuj
>
> ------------------------------
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *darshan b n
> *Sent:* Tuesday, August 12, 2008 12:09 PM
> *To:* [email protected]
> *Subject:* Re: [Sipp-users] Sipp-users Digest, Vol 27, Issue 3
>
>
> i want know the what is the command to excecute sipp with PCAP
>
> Regards
> darshan
>
>
> On 08/08/2008, [EMAIL PROTECTED] <
> [EMAIL PROTECTED]> wrote:
>>
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>>
>> Today's Topics:
>>
>> 1. Re: SUBSCRIBE scenario (??????? ?.?.)
>> 2. SIPp problem with pcap ( Jes?s Camacho Rodr?guez )
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Thu, 07 Aug 2008 18:35:49 +0400
>> From: ??????? ?.?. <[EMAIL PROTECTED]>
>> Subject: Re: [Sipp-users] SUBSCRIBE scenario
>> To: [email protected], ims.asuser ims.asuser"
>> <[EMAIL PROTECTED]>
>> Message-ID: <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset=windows-1251
>>
>>
>> Hi Khaldon,
>>
>> i did perfomance tests of OpenIMSCore using SIPp 2.1 and SIPp IMS Bench.
>> Normal Registration signaling flow looks like:
>>
>> > 1) register ->
>> > 2) <-401
>> > 3) register->
>> > 4) <-200
>>
>> it's enough to provide the registration for user. Pay attention to
>> P-Headers ;)
>>
>> In addition you'll need to fill the FoHSS with a number of subscriber
>> entities, to handle this issue in automatic mode analyse the SQL scripts
>> used for creation of Alice and Bob.
>>
>> Regards,
>> Ivan Kuzmin.
>> State University of Telecomunications, St.Petersburg, Russia.
>>
>>
>> -----Original Message-----
>> From: "ims.asuser ims.asuser" <[EMAIL PROTECTED]>
>> To: [email protected]
>> Date: Thu, 7 Aug 2008 16:16:40 +0200
>> Subject: [Sipp-users] SUBSCRIBE scenario
>>
>> >
>> > Hi all,
>> >
>> > I would like to test the OpenIMSCore platform (I will perform as much as
>> I
>> > can). These tests will be then given to the IMS community.
>> > First of all, did anyone perform OpenIMSCOre tests using SIPp?
>> >
>> > I would like to perform first in different stage:
>> > 1) Registration
>> > 2) Calls
>> > 3) Instant Messaging
>> >
>> > So far, I'm only on the first step (registering a user). I only managed
>> to
>> > do a simple registration (register ->, <-401, register->, <-200). But I
>> have
>> > problems when I want to go further.
>> > In ims, the scenario looks like that:
>> > 1) register ->
>> > 2) <-401
>> > 3) register->
>> > 4) <-200
>> > 5) subscribe->
>> > 6) <-notify
>> > 7) 200->
>> >
>> > The IMS Core is behaving really eerily when I send a subscribe message
>> then
>> > the simulation stops
>> > Is anyone know why is it working like that?
>> >
>> > Attached are my script and wireshark trace.
>> >
>> > Many thanks,
>> > Khaldon
>> >
>> > ATTACHMENT: text/xml (Registerbob1.xml)
>> > ATTACHMENT: application/octet-stream (SIPp_SUBSCRIBE)
>> >
>> >
>> -------------------------------------------------------------------------
>> > This SF.Net email is sponsored by the Moblin Your Move Developer's
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>> > Build the coolest Linux based applications with Moblin SDK & win great
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>> >
>> > _______________________________________________
>> > Sipp-users mailing list
>> > [email protected]
>> > https://lists.sourceforge.net/lists/listinfo/sipp-users
>> >
>> >
>>
>>
>>
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Thu, 7 Aug 2008 20:33:24 +0200
>> From: " Jes?s Camacho Rodr?guez " <[EMAIL PROTECTED]>
>> Subject: [Sipp-users] SIPp problem with pcap
>> To: [email protected]
>> Message-ID:
>> <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi all,
>>
>> I am doing Asterisks performance tests on a pico-itx box. I compiled SIPp
>> with pcap play support in a Debian Etch machine so I could send RTP
>> packages
>> through a recorded pcap file for simulating traffic. I run SIPp in client
>> mode in one machine and I call to a VoIP phone (actually I wanted to call
>> to
>> a SIPp in server mode but I am calling to a phone for debugging purposes).
>> The problem I have is that SIPp in client mode sends the RTP packages to
>> Asterisk (at least that is what SIPp shows in the "Total RTP pckts sent"),
>> but if I activate the RTP debug mode in Asterisk, I see that the
>> communication is only in one direction (communication is established but
>> RTP
>> packages are only sent to the machine with SIPp client). SIP messages
>> between the boxes seems to arrive fine, and the communication after a
>> while
>> closes without problem, telling the SIPp client that it was a Succesful
>> call. First I thought it could be Asterisk, but when I call from another
>> source (one more VoIP phone I have here) the communication is perfect and
>> RTP packages are running in both directions. So I think it can be SIPp. I
>> have used my own xml scenario based on uac_pcap.xml (it has little
>> changes),
>> but even if I run the original uac_pcap, it doesn't work still. Anyway I
>> add
>> my uac_pcap.xml:
>>
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>
>> <!-- This program is free software; you can redistribute it
>> and/or -->
>> <!-- modify it under the terms of the GNU General Public License as
>> -->
>> <!-- published by the Free Software Foundation; either version 2 of the
>> -->
>> <!-- License, or (at your option) any later
>> version. -->
>>
>> <!-- -->
>> <!-- This program is distributed in the hope that it will be
>> useful, -->
>> <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of
>> -->
>> <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See
>> the -->
>> <!-- GNU General Public License for more details.
>> -->
>>
>> <!-- -->
>> <!-- You should have received a copy of the GNU General Public
>> License -->
>> <!-- along with this program; if not, write to
>> the -->
>> <!-- Free Software Foundation,
>> Inc., -->
>> <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
>> -->
>>
>> <!-- -->
>> <!-- Sipp 'uac' scenario with pcap (rtp) play
>> -->
>>
>> <!-- -->
>>
>> <scenario name="UAC with media">
>> <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
>> <!-- generated by sipp. To do so, use [call_id] keyword.
>> -->
>> <send retrans="500" start_rtd="1">
>> <![CDATA[
>>
>> INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>> From: sipp <sip:sipp@
>> [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
>> To: sut <sip:[EMAIL PROTECTED]:[remote_port]>
>> Call-ID: [call_id]
>> CSeq: 1 INVITE
>> Contact: sip:[EMAIL PROTECTED]:[local_port]
>> Max-Forwards: 70
>> Subject: Performance Test
>> Content-Type: application/sdp
>> Content-Length: [len]
>>
>> v=0
>> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>> s=-
>> c=IN IP[local_ip_type] [local_ip]
>> t=0 0
>> m=audio [auto_media_port] RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>
>> ]]>
>> </send>
>>
>> <recv response="100" optional="true" rtd="1" start_rtd="2">
>> </recv>
>>
>> <recv response="180" optional="true" rtd="2">
>> </recv>
>>
>> <!-- By adding rrs="true" (Record Route Sets), the route sets -->
>> <!-- are saved and used for following messages sent. Useful to test -->
>> <!-- against stateful SIP proxies/B2BUAs. -->
>> <recv response="200" crlf="true">
>> </recv>
>>
>> <!-- Packet lost can be simulated in any send/recv message by -->
>> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
>> <send>
>> <![CDATA[
>>
>> ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>> From: sipp <sip:sipp@
>> [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
>> To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
>> Call-ID: [call_id]
>> CSeq: 1 ACK
>> Contact: sip:[EMAIL PROTECTED]:[local_port]
>> Max-Forwards: 70
>> Subject: Performance Test
>> Content-Length: 0
>>
>> ]]>
>> </send>
>>
>> <!-- Play a pre-recorded PCAP file (RTP stream) -->
>> <nop>
>> <action>
>> <exec play_pcap_audio="pcap/call_g711a.pcap"/>
>> </action>
>> </nop>
>>
>> <!-- Pause 3 minutes, which is approximately the duration of the -->
>> <!-- PCAP file -->
>> <pause milliseconds="180000"/>
>>
>> <!-- Play an out of band DTMF '1' -->
>> <nop>
>> <action>
>> <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
>> </action>
>> </nop>
>>
>> <pause milliseconds="1000"/>
>>
>> <!-- The 'crlf' option inserts a blank line in the statistics report. -->
>> <send retrans="500">
>> <![CDATA[
>>
>> BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>> From: sipp <sip:sipp@
>> [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
>> To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
>> Call-ID: [call_id]
>> CSeq: 2 BYE
>> Contact: sip:[EMAIL PROTECTED]:[local_port]
>> Max-Forwards: 70
>> Subject: Performance Test
>> Content-Length: 0
>>
>> ]]>
>> </send>
>>
>> <recv response="200" crlf="true">
>> </recv>
>>
>> <!-- definition of the response time repartition table (unit is ms) -->
>> <ResponseTimeRepartition value="50, 100, 200, 500, 1100, 2100, 3100, 4100,
>> 5100, 6100, 10000"/>
>>
>> <!-- definition of the call length repartition table (unit is ms) -->
>> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>>
>> </scenario>
>>
>>
>> I am quite crazy about the whole problem, it makes me mad that it works
>> using two phones and that it doesn't when I generate the call with SIPp.
>> Maybe I have to add something to the SIP messages in the script that I am
>> not realizing or I don't know...
>>
>> Thanks, greetings,
>> Jes?s
>>
>> PS. I am not using NAT, all the machines and phones are in the same
>> private
>> network.
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>>
>> ------------------------------
>>
>> -------------------------------------------------------------------------
>> This SF.Net email is sponsored by the Moblin Your Move Developer's
>> challenge
>> Build the coolest Linux based applications with Moblin SDK & win great
>> prizes
>> Grand prize is a trip for two to an Open Source event anywhere in the
>> world
>> http://moblin-contest.org/redirect.php?banner_id=100&url=/
>>
>> ------------------------------
>>
>> _______________________________________________
>> Sipp-users mailing list
>> [email protected]
>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>
>>
>> End of Sipp-users Digest, Vol 27, Issue 3
>> *****************************************
>>
>
>
>
> --
> Darshan B N
>
> Thanks & Regards
>
--
Darshan B N
Thanks & Regards
-------------------------------------------------------------------------
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