Dear Rob,
I successfully created the xml file and input in the necessary
information but I am having issues trying to run it. I input "./sipp
-sf custom-uac.xml" and receive the error message, "2014-04-10
03:08:56:985 1397113736.985701: Unable to load or parse
'custom-uac.xml' xml scenario file." I purposely have local ip port set
because when I did netstat -a on the command prompt, multiple ports were
shown and i was not sure which was the proper one. So I'm testing each
port by filling it in where it says local ip port. I knew that the calls
may not have gone properly until the correct port was specified but i
didnt expect for sipp to be unable to load the xml file at all. Any
possible suggestions?
Thankfully,
David
Below is what I have defined in the XML file.
<!-- 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 default 'uac' scenario.
-->
<!--
-->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be
-->
<!-- generated by sipp. To do so, use [call_id] keyword.
-->
<send retrans="500">
<![CDATA[
INVITE sip:[Kamailio]@[10.0.0.160]:[5060] SIP/2.0
Via: SIP/2.0/[transport]
[10.0.0.210]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[Kamailio]@[10.0.0.160]:[5060]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[10.0.0.210]:[local_port]
Max-Forwards: 70
Subject: Performance Test
-->
<!--
-->
<!-- Sipp default 'uac' scenario.
-->
<!--
-->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be
-->
<!-- generated by sipp. To do so, use [call_id] keyword.
-->
<send retrans="500">
<![CDATA[
INVITE sip:[Kamailio]@[10.0.0.160]:[5060] SIP/2.0
Via: SIP/2.0/[transport]
[10.0.0.210]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[Kamailio]@[10.0.0.160]:[5060]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[10.0.0.210]:[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[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets
-->
<!-- are saved and used for following messages sent. Useful to test
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</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" rtd="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:[Kamailio]@[10.0.0.160]:[5060] SIP/2.0
Via: SIP/2.0/[transport]
[10.0.0.210]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[10.0.0.210]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut
<sip:[Kamailio]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[10.0.0.210]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
<!-- against stateful SIP proxies/B2BUAs.
-->
<recv response="200" rtd="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:[Kamailio]@[10.0.0.160]:[5060] SIP/2.0
Via: SIP/2.0/[transport]
[10.0.0.210]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[10.0.0.210]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut
<sip:[Kamailio]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[10.0.0.210]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option
-->
<!-- or by adding a 'milliseconds = "value"' option here.
-->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report.
-->
<send retrans="500">
<![CDATA[
BYE sip:[Kamailio]@[10.0.0.160]:[5060] 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:[Kamailio]@[10.0.0.160]:[5060]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[10.0.0.210]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
<!-- This delay can be customized by the -d command-line option
-->
<!-- or by adding a 'milliseconds = "value"' option here.
-->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report.
-->
<send retrans="500">
<![CDATA[
BYE sip:[Kamailio]@[10.0.0.160]:[5060] SIP/2.0
Via: SIP/2.0/[transport]
[10.0.0.210]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut
<sip:[Kamailio]@[10.0.0.160]:[5060]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[10.0.0.210]:[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="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms)
-->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
-----Original Message-----
From: Rob Day <r...@rkd.me.uk>
To: Munzer,David J <mund...@ufl.edu>
Cc: sipp-users <sipp-users@lists.sourceforge.net>
Sent: Tue, Apr 8, 2014 4:43 pm
Subject: Re: [Sipp-users] Connecting phone system to SIPp
David,
No, there is no performance difference between a built-in scenario and
a scenario loaded from an XMl file - the built-in scenarios are
actually just XML files stored in the compiled binary.
You'll want to use a Unix text editor to edit the file - you may have
used one on Alpine Linux and already be familiar with it, but if not,
nano is probably a good choice (vim and emacs are more widely used,
but have a bigger learning curve). I'm not sure whether this is
installed by default on Cygwin, though - you may need to run the
installer again and choose it.
Best,
Rob
On 8 April 2014 20:54, Munzer,David J <mund...@ufl.edu> wrote:
> Dear Rob,
>
> Will this limit the amount of call flow since my primary use of this
> software is for load testing. In addition, what command do i do to
> edit
> the xml file? Simply typing ./custom-uac.xml does not accomplish
> anything.
>
>
> Thankfully,
> David
>
>
> -----Original Message-----
> From: Rob Day <r...@rkd.me.uk>
> To: Munzer,David J <mund...@ufl.edu>
> Cc: sipp-users <sipp-users@lists.sourceforge.net>
> Sent: Tue, Apr 8, 2014 2:11 pm
> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>
> Hi David,
>
> Do you know the SIP URIs you want to use? If so, you can simply
> create
> a custom scenario file (e.g. by running `sipp -sd uac >
> custom-uac.xml`), edit that custom scenario file to use the right SIP
> URIs in each message, then use that scenario rather than the built-in
> one (by using `-sf custom-uac.xml` instead of `-sn uac`).
>
> Best,
> Rob
>
> On 8 April 2014 19:44, Munzer,David J <mund...@ufl.edu> wrote:
>> Dear Rob,
>>
>> The default scenario is making a call to the address/port of
>> Kamailio.
>> How do I include both the address of the SIP server that I making a
>> call
>> to and the address of one of the actual registered users? The
>> authorization page did not clarify that well.
>>
>> Thankfully,
>> David
>>
>>
>> -----Original Message-----
>> From: Rob Day <r...@rkd.me.uk>
>> To: Munzer,David J <mund...@ufl.edu>
>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>> Sent: Tue, Apr 8, 2014 3:04 am
>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>
>> David,
>>
>> The default scenario makes a call to sip:service@10.0.0.160:5060, as
>> you can see in the error log - is this user configured and
>> registered
>> in Kamailio? (The -s option can change the "service" part, e.g. to a
>> phone number of your choosing.)
>>
>> It may be worth looking at what logs Kamailio has - those should
>> help
>> determine why it returns a 404.
>>
>> Best,
>> Rob
>>
>> On 8 April 2014 03:44, Munzer,David J <mund...@ufl.edu> wrote:
>>> Hi Rob,
>>>
>>> I tried your suggestion of inputting the "./sipp -sn uac -i (My
>>> computer's IP address) (Kamailio's IP address) -trace_err". SIPP is
>>> recognizing the Kamailio server but it fails after sending the
>>> Invite
>>> 100 message. It gave this message, 2: Aborting call on unexpected
>>> message for Call-Id '1-3288@10.0.0.210': while ex 'ecting '100'
>>> (index
>>> 1), received 'SIP/2.0 404 Not Found. I've considered that i may
>>> need
>>> to
>>> run on a different integrated scenario and that the error could
>>> come
>>> from Kamailio not authorizing the call due to me not providing
>>> further
>>> account information. However, if it was this case, I believe I
>>> would
>>> have received a 401 (Unauthorized) or a 407 (Proxy Authentication
>>> Required). If you have any ideas, please let me know. Below is the
>>> attempted call.
>>>
>>> Thankfully, David Munzer
>>>
>>> $ ./sipp -sn uac -i (My computer's IP address) (Kamailio's IP
>>> address)
>>> -trace_err
>>> Warning: open file limit > FD_SETSIZE; limiting max. # of open
>>> files
>>> to
>>> FD_SETSI
>>> ZE = 64
>>> Resolving remote host '10.0.0.160'... Done.
>>> ------------------------------ Scenario Screen -------- [1-9]:
>>> Change
>>> Screen --
>>> Call-rate(length) Port Total-time Total-calls Remote-host
>>> 10.0(0 ms)/1.000s 5060 10.60 s 106
>>> 10.0.0.160:5060(UDP)
>>>
>>> 0 new calls during 0.000 s period 0 ms scheduler resolution
>>> 0 calls (limit 30) Peak was 1 calls, after 0
>>> s
>>> 0 Running, 109 Paused, 0 Woken up
>>> 0 dead call msg (discarded) 0 out-of-call msg
>>> (discarded)
>>> 1 open sockets
>>>
>>> Messages Retrans Timeout
>>> Unexpected-Msg
>>> INVITE ----------> 106 0 0
>>> 100 <---------- 0 0 0 106
>>> 180 <---------- 0 0 0 0
>>> 183 <---------- 0 0 0 0
>>> 200 <---------- E-RTD1 0 0 0 0
>>> ACK ----------> 0 0
>>> Pause [ 0ms] 0 0
>>> BYE ----------> 0 0 0
>>> 200 <---------- 0 0 0 0
>>>
>>> ------------------------------ Test Terminated
>>> --------------------------------
>>>
>>>
>>> ----------------------------- Statistics Screen ------- [1-9]:
>>> Change
>>> Screen --
>>> Start Time | 2014-04-05 02:20:42:441
>>> 1396678842.441802
>>> Last Reset Time | 2014-04-05 02:20:53:081
>>> 1396678853.081802
>>> Current Time | 2014-04-05 02:20:53:082
>>> 1396678853.082802
>>>
>>>
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Counter Name | Periodic value | Cumulative
>>> value
>>>
>>>
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Elapsed Time | 00:00:00:001 |
>>> 00:00:10:641
>>> Call Rate | 0.000 cps | 9.961
>>> cps
>>>
>>>
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Incoming call created | 0 | 0
>>> OutGoing call created | 0 | 106
>>> Total Call created | | 106
>>> Current Call | 0 |
>>>
>>>
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Successful call | 0 | 0
>>> Failed call | 0 | 106
>>>
>>>
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Response Time 1 | 00:00:00:000 |
>>> 00:00:00:000
>>> Call Length | 00:00:00:000 |
>>> 00:00:00:004
>>> ------------------------------ Test Terminated
>>> --------------------------------
>>>
>>> 2014-04-05 02:20:53:077 1396678853.077802: Aborting call on
>>> unexpected m
>>> essage for
>>> Call-Id
>>> '106-3288@10.0.0.210': while expecting '100' (index 1), recei
>>>
>>> ved 'SIP/2.0 404 Not Found
>>> Via: SIP/2.0/UDP
>>> 10.0.0.210:5060;branch=z9hG4bK-3288-106-0;rport=5060
>>> From: sipp <sip:sipp@10.0.0.210:5060>;tag=3288SIPpTag00106
>>> To: sut
>>>
>>>
>>>
>>> <sip:service@10.0.0.160:5060>;tag=fc4b70b0517cb156b1fb39a76698f743-5763
>>> Call-ID: 106-3288@10.0.0.210
>>> CSeq: 1 INVITE
>>> Server: kamailio (4.0.4 (i386/linux))
>>> Content-Length: 0
>>>
>>> '.
>>> sipp: There were more errors, see 'uac_3288_errors.log' file
>>>
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: Rob Day <r...@rkd.me.uk>
>>> To: davidjmunzer <davidjmun...@aol.com>
>>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>>> Sent: Wed, Apr 2, 2014 2:24 pm
>>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>>
>>> David,
>>>
>>> I think that is the wrong syntax - if you have Kamailio's IP
>>> address
>>> after the -i, SIPp will try to bind that IP address (which it
>>> doesn't
>>> own) and fail. You need "./sipp -sn uac -i <your local network IP
>>> address> <Kamailio machine IP address>"
>>>
>>> Note that you need to use the IP address that can talk to your
>>> network
>>> (the one which Kamailio is on, probably 192.168.x.x), not 127.0.0.1
>>> (which is localhost-only).
>>>
>>> Best,
>>> Rob
>>>
>>> On 2 April 2014 20:22, <davidjmun...@aol.com> wrote:
>>>> Hi Rob,
>>>>
>>>> I tried doing that by imputing ./sipp -sn uac 127.0.0.1 -i IP
>>>> dress,
>>>> It
>>>> responds with the error message, 1396509142.105827: Unable to bind
>>>> main
>>>> socket, errno = 125 (Cannot assign requested address). Is there an
>>>> issue
>>>> with my syntax, since I don't see why SIPP shouldn't be able to
>>>> access
>>>> Kamailio's IP address.
>>>>
>>>> Thankfully
>>>> David
>>>>
>>>>
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Rob Day <r...@rkd.me.uk>
>>>> To: davidjmunzer <davidjmun...@aol.com>
>>>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>>>> Sent: Wed, Apr 2, 2014 12:54 pm
>>>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>>>
>>>> Hi David,
>>>>
>>>> I think this may be because your Windows machine provides its IPv6
>>>> or
>>>> its localhost address first, so SIPp uses that and is then unable
>>>> to
>>>> send messages to other IPv4 machines on the network. If you
>>>> explicitly
>>>> specify an IP address to bind to (with the -i option) you should
>>>> get
>>>> better results.
>>>>
>>>> Best,
>>>> Rob
>>>>
>>>> On 2 April 2014 19:18, <davidjmun...@aol.com> wrote:
>>>>> The SIP server that I am using is Kamailio.
>>>>>
>>>>>
>>>>> -----Original Message-----
>>>>> From: Rob Day <r...@rkd.me.uk>
>>>>> To: Munzer,David J <mund...@ufl.edu>
>>>>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>>>>> Sent: Wed, Mar 26, 2014 1:14 pm
>>>>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>>>>
>>>>> Rob,
>>>>>
>>>>> By phone system, I do mean SIP server, specifically a combination
>>>>> of
>>>>> Kamailio and Freeswitch. When I try running the program using
>>>>> "./sipp -sn
>>>>> uac the ip address", It informs me that it's unable to send UDP
>>>>> message:
>>>>> Bad address. I've checked that the SIP server's address is
>>>>> correct
>>>>> by
>>>>> doing
>>>>> ip add on the SIP server to verify the IP address. Any ideas how
>>>>> to
>>>>> approach this issue?
>>>>>
>>>>> Thankfully,
>>>>> David
>>>>>
>>>>> David,
>>>>>
>>>>> When you say that you have a phone system running, do you mean
>>>>> that
>>>>> you have a SIP server (Kamailio/Clearwater/Asterisk) running, or
>>>>> something else?
>>>>>
>>>>> If you have a SIP server, it is probably listening on port 5060
>>>>> (though you can check by running `netstat -lnp`) and you can just
>>>>> give
>>>>> the IP address of that machine as a command-line argument to
>>>>> SIPp.
>>>>> I'm
>>>>> assuming you want to use SIPp in UAC mode to test this phone
>>>>> system
>>>>> -
>>>>> if you want SIPp in UAS mode, handling calls sent to it by that
>>>>> SIP
>>>>> server, you'll need to check the documentation of that SIP server
>>>>> to
>>>>> find how to configure it.
>>>>>
>>>>> SIPp only communicates through the SIP protocol, so if by 'phone
>>>>> system' you don't mean a SIP server, you'll have to set one up to
>>>>> translate between SIP and whatever phone system you have.
>>>>>
>>>>> Best,
>>>>> Rob
>>>>>
>>>>> On 25 March 2014 19:14, Munzer,David J <mund...@ufl.edu> wrote:
>>>>>> Hi,
>>>>>>
>>>>>> I have just finished installing SIPp and am not sure how to
>>>>>> connect
>>>>>> my
>>>>>> phone system to SIPp. My computer is running the program on
>>>>>> Windows
>>>>>> 7
>>>>>> through Cygwin. My phone system runs on Alpine Linuz through a
>>>>>> USB.
>>>>>> Because of the two different operating systems, I need need to
>>>>>> connect
>>>>>> the two most likely through the IP address. However, I am unsure
>>>>>> how to
>>>>>> go about this. I would really appreciate help on this matter.
>>>>>>
>>>>>> Thankfully,
>>>>>> David
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------------
>>>>>> Learn Graph Databases - Download FREE O'Reilly Book
>>>>>> "Graph Databases" is the definitive new guide to graph databases
>>>>>> and
>>>>>> their
>>>>>> applications. Written by three acclaimed leaders in the field,
>>>>>> this first edition is now available. Download your free book
>>>>>> today!
>>>>>> http://p.sf.net/sfu/13534_NeoTech
>>>>>> _______________________________________________
>>>>>> Sipp-users mailing list
>>>>>> Sipp-users@lists.sourceforge.net
>>>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> Sipp-users mailing list
>>>>> Sipp-users@lists.sourceforge.net
>>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Sipp-users mailing list
>>>> Sipp-users@lists.sourceforge.net
>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>
>>>
>>>
>>>
>>>
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>>
>>
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>
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