SIPpSIPp reference documentationForewordThis version of the documentation is for SIPp 3.3 branch and describes some features not present in earlier versions. See the sidebar to access documentation for previous versions.
SIPp is a performance testing tool for the SIP protocol. It includes a
few basic SipStone user agent scenarios (UAC and UAS) and establishes and
releases multiple calls with the INVITE and BYE methods. It can also reads
XML scenario files describing any performance testing configuration. It
features the dynamic display of statistics about running tests (call rate,
round trip delay, and message statistics), periodic CSV statistics dumps,
TCP and UDP over multiple sockets or multiplexed with retransmission
management, regular expressions and variables in scenario files, and
dynamically adjustable call rates.
SIPp can be used to test many real SIP equipements like SIP proxies,
B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very
useful to emulate thousands of user agents calling your SIP system.
Want to see it?
Here is a screenshot
And here is a video (Windows Media Player 9 codec or above required) of
SIPp in action:
sipp-01.wmv
InstallationGetting SIPp
SIPp is released under the
GNU GPL license.
All the terms of the license apply. It was originally created and provided to the SIP
community by Hewlett-Packard
engineers in hope it can be useful, but HP does not
provide any support nor warranty concerning SIPp.
Stable release
Like many other "open source" projects, there are two versions of
SIPp: a stable and unstable release. Stable release: before being
labelled as "stable", a SIPp release is thoroughly tested. So you
can be confident that all mentioned features will work :)
Use the stable release for your everyday use and if you are
not blocked by a specific feature present in the "unstable release"
(see below).
SIPp
stable download page
Unstable release
Unstable release: all new features and bug fixes are checked in
SIPp's
SVN repository as soon as they are available.
Use the unstable release if you absolutely need a bug fix or
a feature that is not in the stable release. Available platforms
SIPp is available on almost all UNIX platforms: HPUX, Tru64,
Linux (RedHat, Debian, FreeBSD), Solaris/SunOS.
A Windows port has been contributed. You can now compile SIPp under
Cygwin.
SIPp works only on Windows XP and later versions and will not
work on Win2000. This is because of IPv6 support.Installing SIPp
On Linux, SIPp is provided in the form of source code. You will need to
compile SIPp to actually use it.
Pre-requisites to compile SIPp are :
C++ Compiler
curses or ncurses library
For TLS support: OpenSSL >= 0.9.8
For pcap play support: libpcap and libnet
For SCTP support: lksctp-tools
For distributed pauses: Gnu Scientific Libraries
You have four options to compile SIPp:
Without TLS (Transport Layer Security), SCTP or PCAP support:
This is the recommended setup if you don't need to handle SCTP, TLS or PCAP. In this case, there are no dependencies to install before building SIPp. It is straightforward:
With TLS support, you must have
installed OpenSSL library (>=0.9.8)
(which may come with your system). Building SIPp consists only in adding
the "--with-openssl" option to the configure command:
With PCAP play support:
With SCTP support:
You can also combine these various options, e.g.::
To enable GSL at compile time,
you must install GSL and its include files, as well as un-comment
the lines in the local.mk file of SIPp distribution. Then, re-compile SIPp.
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6
extension for CYGWIN (http://win6.jp/Cygwin/),
as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.
To compile SIPp on Windows with pcap (media support), you must:
Copy the WinPcap developer package to "C:\cygwin\lib\WpdPack"
Remove or rename "pthread.h" in "C:\cygwin\lib\WpdPack\Include", as it interfers with pthread.h from cygwin
Compile according to the instructions above.
Increasing File Descriptors Limit
If your system does not supports enough file descriptors,
you may experience problems when using the TCP/TLS mode with many simultaneous calls.
You have two ways to overcome this limit: either use the -max_socket
command line option or change the limits of your system.
Depending on the operating system you use, different procedures
allow you to increase the maximum number of file descriptors:
On Linux 2.4 kernels the default number of file descriptors can
be increased by modifying the /etc/security/limits.conf
and the /etc/pam.d/login file.
Open the /etc/security/limits.conf file and add the following lines:
Open the /etc/pam.d/login and add the following line
The system file descriptor limit is set in the /proc/sys/fs/file-max file.
The following command will increase the file descriptor limit:
/proc/sys/fs/file-max]]>
To increase the number of file descriptors to its maximum limit
(65535) set in the /etc/security/limits.conf file, type:
Logout then login again to make the changes effective.
On HP-UX systems the default number of file descriptors
can be increased by modifying the system configuration with the sam utility.
In the Kernel Configuration menu, select Configurable parameters,
and change the following attributes:
Using SIPpMain features
SIPp allows to generate one or many SIP calls to one remote
system. The tool is started from the command line. In this example,
two SIPp are started in front of each other to demonstrate SIPp
capabilities.
Run sipp with embedded server (uas) scenario:
On the same host, run sipp with embedded client (uac) scenario
Integrated scenarios
Integrated scenarios? Yes, there are scenarios that are embedded
in SIPp executable. While you can create your own custom SIP
scenarios (see how to create your own XML
scenarios), a few basic (yet useful) scenarios are available
in SIPp executable.
Scenario file: ooc_default.xml (original XML file)
When a SIPp UAC receives an out-of-call request, it instantiates an out-of-call scenario. By default this scenario simply replies with a 200 OK response. This scenario can be overridden by passing the -oocsf or -oocsn command line options.
|
]]>3PCC
3PCC stands for 3rd Party Call Control. 3PCC is described in
RFC 3725.
While this feature was first developped to allow 3PCC like scenarios,
it can also be used for every case where you would need one SIPp to talk
to several remotes.
In order to keep SIPp simple (remember, it's a test tool!),
one SIPp instance can only talk to one remote. Which is an issue
in 3PCC call flows, like call flow I (SIPp beeing a controller):
The 3PCC feature in SIPp allows to have two SIPp instances
launched and synchronised together. If we take the example of
call flow I, one SIPp instance will take care of the dialog with
remote A (this instance is called 3PCC-C-A for 3PCC-Controller-A-Side)
and another SIPp instance will take care of the dialog with remote B
(this instance is called 3PCC-C-B for 3PCC-Controller-B-Side).
The 3PCC call flow I will, in reality, look like this
(Controller has been divided in two SIPp instances):
As you can see, we need to pass information
between both sides of the controller. SDP "offer1" is provided
by A in message (2) and needs to be sent to B side in message (3).
This mechanism is implemented
in the scenarios through the <sendCmd> command. This:
]]>
Will send a "command" to the twin SIPp instance. Note that including
the Call-ID is mandatory in order to correlate the commands to
actual calls. In the same manner, this:
]]>
Will receive a "command" from the twin SIPp instance.
Using the regular expression mechanism,
the content is retrieved
and stored in a call variable ($2 in this case), ready to be
reinjected
;tag=[call_number]
To: sut [peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
[$2]
]]]]>
]]>
In other words, sendCmd and recvCmd can be seen as synchronization points
between two SIPp instances, with the ability to pass parameters
between each other.
Another scenario that has been reported to be do-able with the
3PCC feature is the following:
A calls B. B answers. B and A converse
B calls C. C answers. C and B converse
B "REFER"s A to C and asks to replace A-B call with B-C call.
A accepts. A and C talk. B drops out of the calls.
3PCC Extended
An extension of the 3pcc mode is implemented in sipp.
This feature allows n twin sipp instances to communicate each other,
each one of them being connected to a remote host.
The sipp instance which initiates the call is launched in "master" mode.
The others are launched in "slave" mode. Twin sipp instances have names,
given in the command line (for example, s1, s2...sn for the slaves and m for the master)
Correspondances between instances names and their addresses must be stored in a file (provided by -slave_cfg command line
argument), in the following format:
Each twin sipp instance must access a different copy of this file.
sendCmd and recvCmd have additional attributes:
]]>
Will send a command to the "s1" peer instance, which can be either master or slave,
depending on the command line argument, which must be consistent with the scenario:
a slave instance cannot have a sendCmd action before having any recvCmd.
Note that the message must contain a "From" field, filled with the name of the sender.
]]>
Indicates that the twin command is expected to be received from the "m" peer instance.
Note that the master must be the launched at last.
There is no integrated scenarios for the 3pcc extended mode, but you can easily adapt those from 3pcc.
Example: the following drawing illustrate the entire procedure. The arrows that are
shown between SIPp master and slaves depict only the synchronization commands exchanged between
the different SIPp instances. The SIP message exchange takes place as usual.
Controlling SIPp
SIPp can be controlled interactively through the keyboard or via a
UDP socket. SIPp supports both 'hot' keys that can be entered at any time
and also a simple command mode. The hot keys are:
Key
Action
+
Increase the call rate by 1 * rate_scale
*
Increase the call rate by 10 * rate_scale
-
Decrease the call rate by 1 * rate_scale
/
Decrease the call rate by 10 * rate_scale
c
Enter command mode
q
Quit SIPp (after all calls complete, enter a second time to quit immediately)
Q
Quit SIPp immediately
s
Dump screens to the log file (if -trace_screen is passed)
p
Pause traffic
1
Display the scenario screen
2
Display the statistics screen
3
Display the repartition screen
4
Display the variable screen
5
Display the TDM screen
6-9
Display the second through fifth repartition screen.
In command mode, you can type a single line command that instructs
SIPp to take some action. Command mode is more versatile than the hot keys,
but takes more time to input some common actions. The following commands
are available:
List of Interactive Commands
Command
Description
Example
dump tasks
Prints a list of active tasks (most tasks are calls) to the error log.
dump tasks
set rate X
Sets the call rate.
set rate 10
set rate-scale X
Sets the rate scale, which adjusts the speed of '+', '-', '*', and '/'.
set rate-scale 10
set users X
Sets the number of users (only valid when -users is specified).
set rate 10
set limit X
Sets the open call limit (equivalent to -l option)
set limit 100
set hide <true|false>
Should the hide XML attribute be respected?
set hide false
set index <true|false>
Display message indexes in the scenario screen.
set index true
set display <main|ooc>
Changes the scenario that is displayed to either the main or the out-of-call scenario.
set display main set display ooc
trace <log> <on|off>
Turns log on or off at run time. Valid values for log are "error", "logs", "messages", and "shortmessages".
trace error on
Traffic control
SIPp generates SIP traffic according to the scenario specified. You
can control the number of calls (scenario) that are started per second.
If you pass the -users option, then you need to control
the number of instantiated users. You can control the rate through:
Interactive hot keys (described in the previous section)
Interactive Commands
Startup Parameters
There are two
commands that control rates: set rate X sets the current
call rate to X. Additionally, set rate-scale X sets the
rate_scale parameter to X. This enables you to use the '+', '-', '*',
and '/' keys to set the rate more quickly. For example, if you do
set rate-scale 100, then each time you press '+', the call
rate is increased by 100 calls and each time you press '*', the call
rate is increased by 1000 calls. Similarly, for a user based benchmark
you can run set users X.
At starting time, you can control the rate by specifying parameters
on the command line:
"-r" to specify the call rate in number of calls per seconds
"-rp" to specify the "rate period"
in milliseconds for the call rate (default is 1000ms/1sec).
This allows you to have n calls every m milliseconds (by using -r n -rp m).
Example: run SIPp at 7 calls every 2 seconds (3.5 calls per second)
You can also pause the traffic by pressing the 'p' key.
SIPp will stop placing new calls and wait until all current calls go to their end.
You can resume the traffic by pressing 'p' again.
To quit SIPp, press the 'q' key.
SIPp will stop placing new calls and wait until all current calls go to their end.
SIPp will then exit.
You can also force SIPp to quit immediatly by pressing the 'Q' key.
Current calls will be terminated by sending a BYE or CANCEL message (depending if the calls have been established or not).
The same behaviour is obtained by pressing 'q' twice.
TIP: you can place a defined number of calls and
have SIPp exit when this is done. Use the -m option on the
command line.Remote control
SIPp can be "remote-controlled" through a UDP socket. This allows for example
To automate a series of actions, like increasing the call rate smoothly,
wait for 10 seconds, increase more, wait for 1 minute and loop
Have a feedback loop so that an application under test can
remote control SIPp to lower the load, pause the traffic, ...
Each SIPp instance is listening to a UDP socket. It
starts to listen to port 8888 and each following SIPp instance (up to 60)
will listen to base_port + 1 (8889, 8890, ...).
It is then possible to control SIPp like this:
/dev/udp/x.y.z.t/8888 -> put SIPp in pause state (p key)
echo q >/dev/udp/x.y.z.t/8888 -> quit SIPp (q key)]]>All keys available through keyboard are also available in
the remote control interface
You could also have a small shell script to automate a serie of action.
For example, this script will
increase the call rate by 10 more new calls/s every 5 seconds, wait at this call rate
for one minute and exit SIPp:
To send a command to SIPp, preface it with 'c'. For example: echo "cset rate 100" >/dev/udp/127.0.0.1/8888 sets the call rate to 100.
Running SIPp in background
SIPp can be launched in background mode (-bg command
line option).
By doing so, SIPp will be detached from the current terminal and run
in the background. The PID of the SIPp process is provided. If you didn't specify a number of calls to execute
with the -m option, SIPp will run forever.
There is a mechanism implemented to stop SIPp smoothly. The command
kill -SIGUSR1 [SIPp_PID] will instruct SIPp to stop placing
any new calls and finish all ongoing calls before exiting.
When using the background mode, the main sipp instance stops and a child process will continue the job. Therefore,
the log files names will contain another PID than the actual sipp instance PID.
Create your own XML scenarios
Of course embedded scenarios will not be enough. So it's time to
create your own scenarios. A SIPp scenario is written in XML
(a DTD that may help you write SIPp
scenarios does exist and has been tested with jEdit - this is described in a later section).
A scenario will always start with:
]]>
And end with:
]]>
Easy, huh? Ok, now let's see what can be put inside. You are not
obliged to read the whole table now! Just go in the next section for
an example.
There are many common attributes used for flow control and
statistics, that can be used for all of the message commands (i.e.,
<send>, <recv>,
<nop>, <pause>,
<sendCmd>, and <recvCmd>).
List of attributes common to all commands
Attribute(s)
Description
Example
start_rtd
Starts one of the "Response Time Duration" timer.
(see statistics section).
<send start_rtd="invite">: the timer named "invite" will start when the message is sent.
rtd
Stops one of the 5 "Response Time Duration" timer.
<send rtd="2">: the timer number 2 will stop when the message is sent.
repeat_rtd
Used with a rtd attribute, it allows the corresponding "Response Time Duration" timer
to be counted more than once per call (useful for loop call flows).
<send rtd="1" repeat_rtd="true">: the timer number 1 value will be printed but the timer won't stop.
crlf
Displays an empty line after the arrow for the message in main SIPp screen.
<send crlf="true">
next
You can put a "next" in any command element to go to another part of the script when you are done with sending the message. For optional receives, the next is only taken if that message was received. See conditional branching section for more info.
Example to jump to label "12" after sending an ACK:
]]>
Example to jump to label "5" when receiving a 403 message:
]]>
test
You can put a "test" next to a "next" attribute to indicate
that you only want to branch to the label specified with "next"
if the variable specified in "test" is set (through regexp
for example).
See conditional branching section for more info.
Example to jump to label "6" after sending an ACK only if
variable 4 is set:
]]>
chance
In combination with "test", probability to actually branch to another part
of the scenario. Chance can have a value between 0 (never) and 1 (always).
See conditional branching section for more info.
]]>90% chance to go to label "5" if variable "3" is set.
condexec
Executes an element only if the variable in the condexec attribute is set. This attribute allows you to write complex XML scenarios with fewer next attributes and labels.
<nop condexec="executethis">
condexec_inverse
If condexec is set, condexec_inverse inverts the condition in condexec. This allows you to execute an element only when a variable is not set.
<nop condexec="skipthis" condexec_inverse="true">
counter
Increments the counter given as parameter when the message is sent.
The counters are saved in the statistic file.
<send counter="MsgA">: Increments counter "MsgA" when the message is sent.
Each command also has its own unique attributes, listed here:
List of commands with their attributes
Command
Attribute(s)
Description
Example
<send>
retrans
Used for UDP transport only: it specifies the T1 timer value,
as described in SIP RFC 3261, section 17.1.1.2.
<send retrans="500">: will initiate T1 timer to 500 milliseconds (RFC3261 default).
lost
Emulate packet lost. The value is specified as a percentage.
<send lost="10">: 10% of the message sent are actually not sent :).
start_txn
Records the branch ID of this sent message so that responses can be properly matched (without this element the transaction matching is done based on the CSeq method, which is imprecise).
<send start_txn="invite">: Stores the branch ID of this message in the transaction named "invite".
ack_txn
Indicates that the ACK being sent corresponds to the transaction started by a start_txn attribute. Every INVITE with a start_txn tag must have a matching ACK with an ack_txn attribute.
<send ack_txn="invite">: References the branch ID of the transaction named "invite".
<recv>
response
Indicates what SIP message code is expected.
<recv response="200">: SIPp will expect a SIP message with code "200".
request
Indicates what SIP message request is expected.
<recv request="ACK">: SIPp will expect an "ACK" SIP message.
optional
Indicates if the message to receive is optional. In case of an optional
message and if the message is actually received, it is not seen as a unexpected message.
When an unexpected message is received, Sipp looks if this message matches an optional message defined in the previous step of the scenario.
If optional is set to "global", Sipp will look every previous steps of the scenario.
<recv response="100" optional="true">: The 100 SIP message can be received without
being considered as "unexpected".
rrs
Record Route Set. if this attribute is set to "true",
then the "Record-Route:" header of the message received is stored and can be recalled using the [routes] keyword.
<recv response="100" rrs="true">.
auth
Authentication. if this attribute is set to "true",
then the "Proxy-Authenticate:" header of the message received is stored and is used to build
the [authentication] keyword.
<recv response="407" auth="true">.
lost
Emulate packet lost. The value is specified as a percentage.
<recv lost="10">: 10% of the message received are thrown away.
timeout
Specify a timeout while waiting for a message. If the message is not received, the call is aborted, unless an ontimeout label is defined.
<recv timeout="100000">
ontimeout
Specify a label to jump to if the timeout popped before the message to be received.
Example to jump to label "5" when not receiving a 100 message after 100 seconds:
]]>
action
Specify an action when receiving the message. See Actions section for possible actions.
Example of a "regular expression" action:
]]>
regexp_match
Boolean. Indicates if 'request' ('response' is not available) is given as a regular expression. If so, the recv
command will match against the regular expression. This allows to catch several cases
in the same receive command.
Example of a recv command that matches MESSAGE or PUBLISH or SUBSCRIBE requests:
]]>
response_txn
Indicates that this is a response to a transaction that was previously started. To match, the branch ID of the first via header must match the stored transaction ID.
<recv response="200" response_txn="invite" />: Matches only responses to the message sent with start_txn="invite" attribute.
<pause>
milliseconds
Specify the pause delay, in milliseconds. When this delay is not set, the value of the -d command
line parameter is used.
<pause milliseconds="5000"/>: pause the scenario for 5 seconds.
variable
Indicates which call variable to use to determine the length of the pause.
<pause variable="1" /> pauses for the number of milliseconds specified by call variable 1.
distribution
Indicates which statistical distribution to use to determine the length of the pause. Without GSL, you may use uniform or fixed. With GSL, normal, exponential, gamma, lambda, lognormal, negbin, (negative binomial), pareto, and weibull are available. Depending on the distribution you select, you must also supply distribution specific parameters.
The following examples show the various types of distributed pauses:
<pause distribution="fixed" value="1000" /> pauses for 1 second.
<pause distribution="uniform" min="2000" max="5000"/> pauses between 2 and 5 seconds.
The remaining distributions require GSL. In general The
parameter names were chosen to be as consistent with
Wikipedia's distribution description pages.
<pause distribution="normal" mean="60000" stdev="15000"/> provides a normal pause with a mean of 60 seconds (i.e. 60,000 ms) and a standard deviation of 15 seconds. The mean and standard deviation are specified as integer milliseconds. The distribution will look like:
<pause distribution="lognormal" mean="12.28" stdev="1" /> creates a distribution's whose natural logarithm has a mean of 12.28 and a
standard deviation of 1. The mean and standard deviation are specified as
double values (in milliseconds). The distribution will look like:
<pause distribution="exponential" mean="900000"/> creates an exponentially distributed pause with a mean of 15 minutes. The distribution will look like:
<pause distribution="weibull" lambda="3" k ="4"/> creates a Weibull distribution with a scale of 3 and a shape of 4 (see Weibull on Wikipedia for a description of the distribution).
<pause distribution="pareto" k="1" x_m="2"/> creates a Pareto distribution with k and xm of 1 and 2, respectively (see Pareto on Wikipedia for a description of the distribution).
<pause distribution="gamma" k="3" theta="2"/> creates a Gamma distribution with k and theta of 9 and 2, respectively (see Gamma on Wikipedia for a description of the distribution).
<pause distribution="negbin" p="0.1" n="2"/> creates a Negative binomial distribution with p and n of 0.1 and 2, respectively (see Negative Binomial on Wikipedia for a description of the distribution).
sanity_check
By default, statistically distributed pauses are sanity checked to ensure that their 99th percentile values are less than INT_MAX. Setting sanity_check to false disables this behavior.
<pause distribution="lognormal" mean="10" stdev="10" sanity_check="false"/> disables sanity checking of the lognormal distribution.
<nop>
action
The nop command doesn't do anything at SIP level. It is
only there to specify an action to execute. See Actions section for possible actions.
Execute the play_pcap_audio/video action:
]]>
<sendCmd>
<![CDATA[]]>
Content to be sent to the twin 3PCC SIPp instance. The Call-ID must be included in the CDATA. In 3pcc extended mode, the From must be included to.
]]>
dest
3pcc extended mode only: the twin sipp instance which the command will be sent to
<sendCmd dest="s1">: the command will be sent to the "s1" twin instance
<recvCmd>
action
Specify an action when receiving the command. See Actions section for possible actions.
Example of a "regular expression" to retrieve what has been send by a sendCmd command:
]]>
src
3pcc extended mode only: indicate the twin sipp instance which the command is expected to be received from
<recvCmd src = "s1">: the command will be expected to be received from the "s1" twin instance
<label>
id
A label is used when you want to branch to specific parts
in your scenarios. The "id" attribute is an integer where the maximum value is 19.
See conditional branching section for more info.
Example: set label number 13:]]>
<Response Time Repartition>
value
Specify the intervals, in milliseconds, used to distribute the values of response times.
<ResponseTimeRepartition value="10, 20,
30"/>: response time values are distributed
between 0 and 10ms, 10 and 20ms, 20 and 30ms, 30 and
beyond.
<Call Length Repartition>
value
Specify the intervals, in milliseconds, used to distribute the values of the call length measures.
<CallLengthRepartition value="10, 20,
30"/>: call length values are distributed between
0 and 10ms, 10 and 20ms, 20 and 30ms, 30 and beyond.
<Globals>
variables
Specify the name of globally scoped variables.
<Globals variables="foo,bar" />.
<User>
variables
Specify the name of user-scoped variables.
<User variables="foo,bar" />.
<Reference>
variables
Suppresses warnings about unused variables.
<Reference variables="dummy" />
There are not so many commands: send, recv, sendCmd, recvCmd,
pause, ResponseTimeRepartition, CallLengthRepartition, Globals, User, and Reference. To make
things even clearer, nothing is better than an example...
Structure of client (UAC like) XML scenarios
A client scenario is a scenario that starts with a "send" command. So let's start:
Inside the "send" command, you have to enclose your SIP message
between the "<![CDATA" and the "]]>" tags. Everything between
those tags is going to be sent toward the remote system. You may
have noticed that there are strange keywords in the SIP message,
like [service], [remote_ip], .... Those keywords
are used to indicate to SIPp that it has to do something with
it.
Here is the list:
Keyword list
Keyword
Default
Description
[service]
service
Service field, as passed in the -s service_name
[remote_ip]
-
Remote IP address, as passed on the command line.
[remote_port]
5060
Remote IP port, as passed on the command line. You can
add a computed offset [remote_port+3] to this value.
[transport]
UDP
Depending on the value of -t parameter, this will take the values "UDP" or "TCP".
[local_ip]
Primary host IP address
Will take the value of -i parameter.
[local_ip_type]
-
Depending on the address type of -i parameter (IPv4 or IPv6),
local_ip_type will have value "4" for IPv4 and "6" for IPv6.
[local_port]
Chosen by the system
Will take the value of -p parameter.
You can add a computed offset [local_port+3] to this value.
[len]
-
Computed length of the SIP body. To be used in "Content-Length"
header. You can add a computed offset [len+3] to this value.
[call_number]
-
Index. The call_number starts from "1" and is incremented by 1 for each call.
[cseq]
-
Generates automatically the CSeq number. The initial value is 1 by default. It
can be changed by using the -base_cseq command line option.
[call_id]
-
A call_id identifies a call and is generated by SIPp for each new call. In client mode, it is mandatory
to use the value generated by SIPp in the "Call-ID" header. Otherwise, SIPp will not recognise
the answer to the message sent as being part of an existing call.
Note: [call_id] can be pre-pended with an arbitrary string using '///'. Example: Call-ID: ABCDEFGHIJ///[call_id] - it will still be recognized by SIPp as part of the same call.
[media_ip]
-
Depending on the value of -mi parameter, it is the local IP address for RTP echo.
[media_ip_type]
-
Depending on the address type of -mi parameter (IPv4 or IPv6),
media_ip_type will have value "4" for IPv4 and "6" for IPv6. Useful to build the SDP independently
of the media IP type.
[media_port]
-
Depending on the value of -mp parameter, it set the local RTP echo port number. Default
is none. RTP/UDP packets received on that port are echoed to their sender. You can
add a computed offset [media_port+3] to this value.
[auto_media_port]
-
Only for pcap. To make audio and video ports begin from the value of -mp parameter,
and change for each call using a periodical system, modulo 10000 (which limits to 10000 concurrent RTP sessions for pcap_play)
[last_*]
-
The '[last_*]' keyword is replaced automatically by the
specified header if it was present in the last message
received (except if it was a retransmission). If the header
was not present or if no message has been received, the
'[last_*]' keyword is discarded, and all bytes until the end
of the line are also discarded. If the specified header was
present several times in the message, all occurences are
concatenated (CRLF separated) to be used in place of the
'[last_*]' keyword.
[field0-n file=<filename> line=<number>]
-
Used to inject values from an external CSV file. See
"Injecting values from an external CSV
during calls" section. The optional file and line
parameters allow you to select which of the injection files
specified on the command line to use and which line number
from that file.
[file name=<filename>]
-
Inserts the entire contents of filename into the message. Whitespace,
including carriage returns and newlines at the end of the line in the file
are not processed as with other keywords; thus your file must be formatted
exactly as you would like the bytes to appear in the message.
[timestamp]
-
The current time using the same format as error log messages.
[last_message]
-
The last received message.
[$n]
-
Used to inject the value of call variable number n. See "Actions" section
[authentication]
-
Used to put the authentication header. This field can have parameters, in the following form:
[authentication username=myusername password=mypassword]. If no username is provided,
the value from the -au (authentication username) or -s (service) command line parameter is used. If no password is provided, the value
from -ap command line parameter is used. See "Authentication" section
[pid]
-
Provide the process ID (pid) of the main SIPp thread.
[routes]
-
If the "rrs" attribute in a recv command is set to "true",
then the "Record-Route:" header of the message received is stored
and can be recalled using the [routes] keyword
[next_url]
-
If the "rrs" attribute in a recv command is set to "true",
then the [next_url] contains the contents of the Contact header
(i.e within the '<' and '>' of Contact)
[branch]
-
Provide a branch value which is a concatenation of magic cookie
(z9hG4bK) + call number + message index in scenario. An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message.
[msg_index]
-
Provide the message number in the scenario.
[cseq]
-
Provides the CSeq value of the last request received. This value can be incremented (e.g. [cseq+1] adds 1 to
the CSeq value of the last request).
[clock_tick]
-
Includes the internal SIPp clock tick value in the message.
[sipp_version]
-
Includes the SIPp version string in the message.
[tdmmap]
-
Includes the tdm map values used by the call in the message (see -tdmmap option).
[fill]
-
Injects filler characters into the message. The length of the fill
text is equal to the call variable stored in the
variable=N parameter. By default the text is a sequence
of X's, but can be controlled with the text="text" parameter.
[users]
-
If the -users command line option is specified, then this keyword contains the number of users that are currently instantiated.
[userid]
-
If the -users command line option is specified, then this keyword containst he integer identifier of the current user (starting at zero and ending at [users-1]).
Now that the INVITE message is sent, SIPp can wait for an answer by using the "recv" command.
optional="true"
optional="true"
]]>
100 and 180 messages are optional, and 200 is mandatory.
In a "recv" sequence, there must be one mandatory message.
Creating your own SIPp scenarios is not a big deal.
If you want to see other examples, use the -sd parameter
on the command line to display embedded scenarios.
Structure of server (UAS like) XML scenarios
A server scenario is a scenario that starts with a "recv" command.
The syntax and the list of available commands is the same as for
"client" scenarios.
But you are more likely to use [last_*] keywords in those server
side scenarios. For example, a UAS example will look like:
The answering message, 180 Ringing in this case, is built
with the content of headers received in the INVITE message.
Actions
In a "recv" or "recvCmd" command, you have the possibility to
execute an action. Several actions are available:
Regular expressions (ereg)
Log something in aa log file (log)
Execute an external (system), internal (int_cmd) or
pcap_play_audio/pcap_play_video command (exec)
Manipulate double precision variables using arithmetic
Assign string values to a variable
Compare double precision variables
Jump to a particular scenario index
Store the current time into variables
Lookup a key in an indexed injection file
Verify Authorization credentials
Change a Call's Network Destination
Regular expressions
Using regular expressions in SIPp allows to
Extract content of a SIP message or a SIP header and
store it for future usage (called re-injection)
Check that a part of a SIP message or of an header
is matching an expected expression
Regular expressions used in SIPp are defined per
Posix Extended standard (POSIX 1003.2). If you want to
learn how to write regular expressions, I will recommend
this
regexp tutorial.
Here is the syntax of the regexp action:
regexp action syntax
Keyword
Default
Description
regexp
None
Contains the regexp to use for matching the
received message or header. MANDATORY.
search_in
msg
can have four values: "msg" (try to match against
the entire message); "hdr" (try to match against a specific SIP header); "body" (try to match against the SIP message body); or "var" (try to match against a SIPp string variable).
header
None
Header to try to match against. Only used when
the search_in tag is set to hdr. MANDATORY IF
search_in is equal to hdr.
variable
None
Variable to try to match against. Only used when
the search_in tag is set to var. MANDATORY IF
search_in is equal to var.
case_indep
false
To look for a header ignoring case . Only used when
the search_in tag is set to hdr.
occurence
1
To find the nth occurence of a header. Only used when
the search_in tag is set to hdr.
start_line
false
To look only at start of line. Only used when
the search_in tag is set to hdr.
check_it
false
if set to true, the call is marked as failed if
the regexp doesn't match. Can not be combined with check_it_inverse.
check_it_inverse
false
Inverse of check_it. iff set to true, the call is marked as failed if
the regexp does match. Can not be combined with check_it.
assign_to
None
contain the variable id (integer) or a list of
variable id which will be used to store the
result(s) of the matching process between the regexp
and the message. Those variables can be re-used at
a later time either by using '[$n]' in the scenario
to inject the value of the variable in the messages or
by using the content of the variables for conditional
branching. The first variable in the variable list of
assign_to contains the entire regular
expression matching. The following variables contain the
sub-expressions matching. Example: ]]>If the SIP message contains the line
variable 3 contains "o=user1 53655765 2353687637", variable 4 contains "user1",
variable 5 contains "53655765" and variable 8 contains "2353687637".
Note that you can have several regular expressions
in one action.
The following example is used to:
First action:
Extract the first IPv4 address of the received SIP message
Check that we could actually extract this IP address (otherwise
call will be marked as failed)
Assign the extracted IP address to call variables 1
and 2.
Second action:
Extract the Contact: header of the received SIP message
Assign the extracted Contract: header to variable 6.
]]>Log a message
The "log" action allows you to customize your traces. Messages
are printed in the <scenario file name>_<pid>_logs.log file.
Any keyword is expanded to reflect the value actually used.
Logs are generated only if -trace_logs option is set on
the command line.
Example:
]]>
You can use the alternative "warning" action to log a message to SIPp's error log. For example:
]]>Execute a command
The "exec" action allows you to execute "internal", "external", "play_pcap_audio" or "play_pcap_video" commands.
Internal commands
Internal commands (specified using int_cmd attribute) are stop_call, stop_gracefully (similar to pressing 'q'), stop_now (similar to ctrl+C).
Example that stops the execution of the script on receiving a 603 response:
]]>External commands
External commands (specified using command attribute) are anything that can be executed on local host with a shell.
Example that execute a system echo for every INVITE received:
]]>PCAP (media) commands
PCAP play commands (specified using play_pcap_audio / play_pcap_video attributes)
allow you to send a pre-recorded RTP stream using the pcap library.
Choose play_pcap_audio to send the pre-recorded RTP stream using the "m=audio" SIP/SDP line port as a base for the replay.
Choose play_pcap_video to send the pre-recorded RTP stream using the "m=video" SIP/SDP line port as a base.
The play_pcap_audio/video command has the following format: play_pcap_audio="[file_to_play]" with:
file_to_play: the pre-recorded pcap file to play
The action is non-blocking. SIPp will start a light-weight thread to play the file
and the scenario with continue immediately. If needed, you will need to add a pause
to wait for the end of the pcap play.A known bug means that starting a pcap_play_audio command will end any pcap_play_video command, and vice versa; you cannot play both audio and video streams at once.
Example that plays a pre-recorded RTP stream:
]]>Variable Manipulation
You may also perform simple arithmetic (add, subtract,
multiply, divide) on floating point values. The "assign_to" attribute contains
the first operand, and is also the destination of the resulting value.
The second operand is either an immediate value or stored in a variable, represented by
the "value" and "variable" attributes, respectively.
SIPp supports call variables that take on double-precision floating values.
The actions that modify double variables all write to the
variable referenced by the assign_to parameter.
These variables can be assigned using one of three actions:
assign, sample, or todouble. For assign, the
double precision value is stored in the "value" parameter. The sample action assigns
values based on statistical distributions, and uses the same
parameters as a statistically distributed
pauses. Finally, the todouble command converts the variable referenced by the
"variable" attribute to a double before assigning it.
For example, to assign the value 1.0 to $1 and sample from the normal distribution into $2:
]]>
Simple arithmetic is also possible using the <add>, <subtract>, <multiply>, and <divide> actions, which add, subtract, multiply, and divide the variable referenced by assign_to by the value in value.
For example, the following action modifies variable one as follows:
]]>
Rather than using fixed values, you may also retrieve the second operand from a variable, using the <variable> parameter. For example:
]]>String Variables
You can create string variables by using the <assignstr>
command, which accepts two parameters: assign_to and
value. The value may contain any of the same substitutions
that a message can contain. For example:
]]>
A string variable and a value can be compared using the
<strcmp> action. The result is a double value, that is
less than, equal to, or greater than zero if the variable is lexographically
less than, equal to, or greater than the value. The parameters are assign_to,
variable, and value. For example:
]]>Variable Testing
Variable testing allows you to construct loops and control
structures using call variables. THe test
action takes four arguments: variable which
is the variable that to compare against
value, and assign_to which
is a boolean call variable that the result of the test is
stored in. Compare may be one of the following tests:
equal, not_equal,
greater_than, less_than,
greater_than_equal, or
less_than_equal.
Example that sets $2
to true if $1 is less than 10:
]]>lookup
The lookup action is used for indexed injection files (see indexed injection files). The lookup action takes a file and key as input and produces an integer line number as output. For example the following action extracts the username from an authorization header and uses it to find the corresponding line in users.csv.
]]>Updating In-Memory Injection files
Injection files, particularly when an index is defined can serve as an
in-memory data store for your SIPp scenario. The
<insert> and <replace> actions provide
a method of programmatically updating SIPp's
in-memory version of an injection file (there is presently no
way to update the disk-based version). The insert action takes
two parameters: file and value, and the replace action takes an
additional line value. For example, to inserting a new line can
be accomplished as follows:
]]>
Replacing a line is similar, but a line number must be specified.
You will probably want to use the lookup action to obtain the
line number for use with replace as follows:
]]>Jumping to an Index
You can jump to an arbitrary scenario index using the <jump> action. This can be used to create rudimentary subroutines. The caller can save their index using the [msg_index] substitution, and the callee can jump back to the same place using this action. If there is a special label named "_unexp.main" in the scenario, SIPp will jump to that label whenever an unexpected message is received and store the previous address in the variable named "_unexp.retaddr".
Example that jumps to index 5:
]]>
Example that jumps to the index contained in the variable named _unexp.retaddr:
]]>gettimeofday
The gettimeofday action allows you to get the current time in seconds and microseconds since the epoch. For example:
]]>setdest
The setdest action allows you to change the remote end
point for a call. The parameters are the transport, host, and port to connect
the call to. There are certain limitations baed on SIPp's design: you can not
change the transport for a call; and if you are using TCP then multi-socket
support must be selected (i.e. -t tn must be specified). Also, be
aware that frequently using setdest may reduce SIPp's capacity as name
resolution is a blocking operation (thus potentially causing SIPp to stall
while looking up host names). This example connects to the value specified in the [next_url] keyword.
]]>If you are using setdest with IPv6, you must not use square brackets around the address. These have a special meaning to SIPp, and it will try to interpret your IPv6 address as a variable.
Since the port is specified separately, square brackets are never necessary.verifyauth
The verifyauth action checks the Authorization header in
an incoming message against a provided username and password.
The result of the check is stored in a boolean variable.
This allows you to simulate a server which requires
authorization. Currently only simple MD5 digest
authentication is supported. Before using the verifyauth
action, you must send a challenge. For example:
After receiving the second request, you can extract the username provided and compare it against a list of user names and passwords provided as an injection file, and take the appropriate action based on the result:
]]>Variables
For complex scenarios, you will need to store bits of
information that can be used across messages or even calls.
Like other programming languages, SIPp's XML scenario definition
allows you to use variables for this purpose. A variable in SIPp
is referenced by an alphanumeric name. In past versions of SIPp,
variables names were numeric only; thus in this document and the
embedded scenarios, you are likely to see lots of variables of
the form "1", "2", etc.; although when creating new scenarios you
are encouraged to assign meaningful names to your variables.
Aside from a name, SIPp's variables are also loosely typed. The
type of a variable is not explicitly declared, but is instead
inferred from the action that set it. There are four types of variables:
string, regular expression matches, doubles, and booleans. All
mathematical operations take place on doubles. The
<test> and
<verifyauth> actions create boolean
values. String variables and regular expression matches are
similar. When a string's value is called for, a regular
expression match can be substituted. The primary difference is
related to the test attribute (see Conditional Branching). If a
string has been defined, a test is evaluated to true.
However, for a regular expression variable, the regular
expression that set it must match for the test to evaluated to true.
Values can be converted to strings using the
<assignstr>
action. Values can be converted to doubles using the <todouble> action.
Variables also have a scope, which is one of global to all
calls, per-user, or the default per-call. A global variable can be used,
for example to store scenario configuration parameters or to keep
a global counter. A user-variable when combined with the
-users option allows you to keep per-user state
across calls (e.g., if this user has already registered).
Finally, the default per-call variables are useful for copying
values from one SIP message to the next or controlling
branching. Variables can be declared globally or per-user using the
following syntax:
]]>
Local variables need not be declared. To prevent programming
errors, SIPp performs very rudimentary checks to ensure that each
variable is used more than once in the scenario (this helps prevent
some typos from turning into hard to debug errors). Unfortunately,
this can cause some complication with regular expression matching. The regular
expression action must assign the entire matched expression to a
variable. If you are only interested in checking the validity of
the expression (i.e. the check_it attribute is set) or in capturing
a sub-expression, you must still assign the entire expression to a
variable. As this variable is likely only referenced once, you must
inform SIPp that you are knowingly using this variable once with
a Reference clause. For example:
]]>Injecting values from an external CSV during calls
You can use "-inf file_name" as a command line parameter
to input values into the scenarios. The first line of the file should
say whether the data is to be read in sequence (SEQUENTIAL), random
order (RANDOM), or in a user based manner (USER). Each line corresponds
to one call and has one or more ';' delimited data fields and they can be
referred as [field0], [field1], ... in the xml scenario file. Example:
Will be read in sequence (first call will use first line,
second call second line). At any place where the keyword
"[field0]" appears in the scenario file, it will be replaced by
either "Sarah", "Bob" or "Fred" depending on the call. At any
place where the keyword "[field1]" appears in the scenario file,
it will be replaced by either "sipphone32" or "sipphone12" or
"sipphone94" depending on the call. At the end of the file, SIPp
will re-start from the beginning. The file is not limited in
size.
You can override the default line selection strategy with the
optional line argument. For example:
Selects the second line in the file (the first line is line zero.
The line parameters support keywords in the argument, so in
conjunction with a lookup action it is possible to select
values based on a key.
The CSV file can contain comment lines. A comment line is
a line that starts with a "#".
As a picture says more than 1000 words, here is one:
Think of the possibilities of this feature. They are huge.
It is possible to use more than one injection file, and is
necessary when you want to select different types of data in different
ways. For example, when running a user-based benchmark, you may have
a caller.csv with "USER" as the first line and a callee.csv
with "RANDOM" as the first line. To specify which CSV file is
used, add the file= parameter to the keyword. For example:
;tag=[pid]SIPpTag00[call_number]
To: sut user <[field0 file="callee.csv"]>
...]]>
Will select the destination user from callee.csv and the sending
user from caller.csv. If no file parameter is specified, then the
first input file on the command line is used by default.
PRINTF Injection files
An extension of the standard injection file is a "PRINTF" injection file. Often, an input file will has a repetitive nature such as:
SIPp must maintain this structure in memory, which can reduce performance for very large injection files. To eliminate this problem, SIPp can automatically generate such a structured file based on one or more template lines. For example:
Has the same logical meaning as the original example, yet
SIPp only needs to store one entry in memory. Each time a line
is used; SIPp will replace %d with the requested line number
(starting from zero). Standard printf format decimal
specifiers can be used. When more than one template line is
available, SIPp cycles through them. This example:
Is equivalent to the following injection file:
The following parameters are used to control the behavior of
printf injection files:
Printf Injection File Parameters
Parameter
Description
Example
PRINTF
How many virtual lines exist in this file.
PRINTF=10, creates 10 virtual lines
PRINTFMULTIPLE
Multiple the virtual line number by this value before generating the substitutions used.
The -infindex option allows you to generate an
index of an injection file. The arguments to -infindex are the
injection file to index and the field number that should be
indexed. For example if you have an injection file that
contains user names and passwords (as the following):
You may want to extract the password for a given user in the
file. To do this efficiently, SIPp must build an index for the
first field (0). Thus you would pass the argument
-infindex users.csv 0 (assuming the file is named
users.csv). SIPp will create an index that contains the
logical entries {"alice" => 0, "bob" => 1, "carol" => 2}. To
extract a particular password, you can use the lookup action to
store the line number into a variable (say $line) and then use
the keyword[field1 line="[$line]"].
Conditional branchingConditional branching in scenarios
It is possible to execute a scenario in a non-linear
way. You can jump from one part of the scenario to another for example
when a message is received or if a call variable is set.
You define a label (in the xml) as <label id="n"/>
Where n is a number between 1 and 19 (we can easily have more if needed).
The label commands go anywhere in the main scenario between other commands.
To any action command (send, receive, pause, etc.) you add a next="n"
parameter, where n matches the id of a label. When it has done the
command it continues the scenario from that label. This part is
useful with optional receives like 403 messages, because it allows
you to go to a different bit of script to reply to it and then
rejoin at the BYE (or wherever or not).
Alternatively, if you add a test="m" parameter to the next,
it goes to the label only if variable [$m] is set. This allows you
to look for some string in a received packet and alter the
flow either on that or a later part of the script. The
evaluation of a test varies based on the type of call
variable. For regular expressions, at least one match must have been found;
for boolean variables the value must be true; and for all others a value must
have been set (currently this only applies to doubles). For more complicated
tests, see the <test> action.
If you add special cases at the end, don’t forget to put
a label at the real end and jump to it at the end of the normal flow.
Example:
The following example corresponds to the embedded 'branchc' (client side) scenario.
It has to run against the embedded 'branchs' (server side) scenario.
Randomness in conditional branching
To have SIPp behave somewhat more like a "normal" SIP client being used by a human,
it is possible to use "statistical branching".
Wherever you can have a conditional branch on a variable being set (test="4"),
you can also branch based on a statistical decision using the attribute
"chance" (e.g. chance="0.90"). Chance can have a value between 0 (never)
and 1 (always). "test" and "chance" can be combined, i.e. only
branching when the test succeeds and the chance is good.
With this, you can have a variable reaction in a given scenario
(e.g.. answer the call or reject with busy), or run around in a
loop (e.g. registrations) and break out of it after some random number of iterations.
SIP authentication
SIPp supports SIP authentication. Two authentication algorithm are supported: Digest/MD5 ("algorithm="MD5"")
and Digest/AKA ("algorithm="AKAv1-MD5"", as specified by 3GPP for IMS).
Enabling authentication is simple. When receiving a 401 (Unauthorized)
or a 407 (Proxy Authentication Required), you must add auth="true"
in the <recv> command to take the challenge into account.
Then, the authorization header can be re-injected in the next message
by using [authentication] keyword.
Computing the authorization header is done through the usage of the
"[authentication]" keyword. Depending on the algorithm ("MD5" or "AKAv1-MD5"),
different parameters must be passed next to the authentication
keyword:
username: username: if no username is specified, the username is taken from the '-au' (authentication username) or '-s' (service) command
line parameter
password: password: if no password is specified, the password is taken from the '-ap' (authentication
password) command line parameter
username: username: if no username is specified, the username is taken from the '-au' (authentication username) or '-s' (service) command
line parameter
aka_K: Permanent secret key. If no aka_K is provided, the "password" attributed is used as aka_K.
aka_OP: OPerator variant key
aka_AMF: Authentication Management Field (indicates the algorithm and key in use)
In case you want to use authentication with a different username/password or aka_K
for each call, you can do this:
Make a CSV like this:
And an XML like this (the [field1] will be substituted with the full auth string, which is the processed as a new keyword):
From:
Contact: ;transport=[transport]
[field1]
Expires: 300
Call-ID: [call_id]
CSeq: 2 REGISTER
Content-Length: 0
]]]]>
]]>
Example:
auth="true"
;tag=[call_number]
To: sut [peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]]]>
;tag=[call_number]
To: sut
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
]]>[authentication username=foouser]
]]>Initialization Stanza
Some complex scenarios require setting appropriate global variables
at SIPp startup. The initialization stanza allows you do do just that.
To create an initialization stanza, simply surround a series of
<nop> and <label> commands with <init> and
</init>. These <nop>s are executed once at SIPp startup.
The variables within the init stanza, except for globals, are not
shared with calls. For example, this init stanza sets $THINKTIME to 1
if it is not already set (e.g., by the -set command line parameter).
]]>Screens
Several screens are available to monitor SIP traffic. You can
change the screen view by pressing 1 to 9 keys on the keyboard.
Key '1': Scenario screen. It displays a call flow of
the scenario as well as some important informations.
Key '2': Statistics screen. It displays the main statistics
counters. The "Cumulative" column gather all statistics, since
SIPp has been launched. The "Periodic" column gives the statistic
value for the period considered (specified by -f frequency command
line parameter).
Key '3': Repartition screen. It displays the distribution
of response time and call length, as specified in the scenario.
Key '4': Variables screen. It displays informations on
actions in scenario as well as scenario variable informations.
Transport modes
SIPp has several transport modes. The default transport mode is
"UDP mono socket".
UDP mono socket
In UDP mono socket mode (-t u1 command line parameter),
one IP/UDP socket is opened between SIPp and the remote. All calls
are placed using this socket.
This mode is generally used for emulating a relation
between 2 SIP servers.
UDP multi socket
In UDP multi socket mode (-t un command line parameter),
one IP/UDP socket is opened for each new call between SIPp and the remote.
This mode is generally used for emulating user agents calling a SIP server.
UDP with one socket per IP address
In UDP with one socket per IP address mode (-t ui command line parameter),
one IP/UDP socket is opened for each IP address given in the inf file.
In addition to the "-t ui" command line parameter, one must indicate which field in the
inf file is to be used as local IP address for this given call. Use "-ip_field <nb>"
to provide the field number.
There are two distinct cases to use this feature:
Client side: when using -t ui for a client, SIPp will originate each call
with a different IP address, as provided in the inf file. In this case,
when your IP addresses are in field X of the inject file,
then you have to use [fieldX] instead of [local_ip] in your UAC XML scenario file.
Server side: when using -t ui for a server, SIPp will bind itself
to all the IP addresses listed in the inf file instead of using 0.0.0.0. This
will have the effect SIPp will answer the request on the same IP on which
it received the request. In order to have proper Contact and Via fields,
a keyword [server_ip] can be used and provides the IP address on
which a request was received. So when using this, you have to replace the [local_ip] in
your UAS XML scenario file by [server_ip].
In the following diagram, the command line for a client scenario will look like:
./sipp -sf myscenario.xml -t ui -inf database.csv -ip_field 2 192.168.1.1
By doing so, each new call will come sequentially from IP 192.168.0.1, 192.168.0.2, 192.168.0.3, 192.168.0.1, ...
This mode is generally used for emulating user agents, using on IP address
per user agent and calling a SIP server.
TCP mono socket
In TCP mono socket mode (-t t1 command line parameter),
one IP/TCP socket is opened between SIPp and the remote. All calls
are placed using this socket.
This mode is generally used for emulating a relation between 2 SIP servers.
TCP multi socket
In TCP multi socket mode (-t tn command line parameter),
one IP/TCP socket is opened for each new call between SIPp and the remote.
This mode is generally used for emulating user agents calling a SIP server.
TCP reconnections
SIPp handles TCP reconnections. In case the TCP socket is lost,
SIPp will try to reconnect. The following parameters on the command line
control this behaviour:
-max_reconnect: Set the maximum number of reconnection attempts.
-reconnect_close true/false: Should calls be closed on reconnect?
-reconnect_sleep int: How long to sleep (in milliseconds) between the close and reconnect?
TLS mono socket
In TLS mono socket mode (-t l1 command line parameter),
one secured TLS (Transport Layer Security) socket is opened between SIPp and the remote. All calls
are placed using this socket.
This mode is generally used for emulating a relation between 2 SIP servers.
When using TLS transport, SIPp will expect to have two files in the current
directory: a certificate (cacert.pem) and a key (cakey.pem).
If one is protected with a password, SIPp will
ask for it.
SIPp supports X509's CRL (Certificate Revocation List). The CRL is
read and used if -tls_crl command line specifies
a CRL file to read.
TLS multi socket
In TLS multi socket mode (-t ln command line parameter),
one secured TLS (Transport Layer Security) socket is opened for each new call between SIPp and the remote.
This mode is generally used for emulating user agents calling a SIP server.
SCTP mono socket
In SCTP mono socket mode (-t s1 command line parameter),
one SCTP (Stream Transmission Control Protocol) socket is opened between SIPp and the remote. All calls
are placed using this socket.
This mode is generally used for emulating a relation between 2 SIP servers.
The -multihome, -heartbeat, -assocmaxret, -pathmaxret, -pmtu and -gracefulclose command-line arguments allow control over specific features of the SCTP protocol, but are usually not necessary.
SCTP multi socket
In SCTP multi socket mode (-t sn command line parameter),
one SCTP socket is opened for each new call between SIPp and the remote.
This mode is generally used for emulating user agents calling a SIP server.
IPv6 support
SIPp includes IPv6 support. To use IPv6, just specify the local
IP address (-i command line parameter) to be an IPv6 IP address.
The following example launches a UAS server listening on port 5063 and a UAC client sending
IPv6 traffic to that port.
The Pcap play feature may currently not work on IPv6.Multi-socket limit
When using one of the "multi-socket" transports, the maximum number of sockets that can be opened
(which corresponds to the number of simultaneous calls) will be determined by
the system (see how to increase file descriptors section to
modify those limits). You can also limit the number of socket used by using the -max_socket
command line option. Once the maximum number of opened sockets is reached,
the traffic will be distributed over the sockets already opened.
Handling media with SIPp
SIPp is originally a signalling plane traffic generator. There is a limited
support of media plane (RTP).
RTP echo
The "RTP echo" feature allows SIPp to listen to one or two local IP
address and port (specified using -mi and
-mp command line parameters) for RTP media. Everything
that is received on this address/port is echoed back to the
sender.
RTP/UDP packets coming on this port + 2 are also echoed to their sender (used for
sound and video echo).
PCAP Play
The PCAP play feature makes use of the PCAP library
to replay pre-recorded RTP streams towards a destination. RTP streams can be
recorded by tools like Wireshark
(formerly known as Ethereal) or tcpdump. This allows you to:
Play any RTP stream (voice, video, voice+video, out of band DTMFs/RFC 2833, T38 fax, ...)
Use any codec as the codec is not handled by SIPp
Emulate precisely the behavior of any SIP equipment as the
pcap play will try to replay the RTP stream as it was recorded (limited
to the performances of the system).
Reproduce exactly what has been captured using an IP sniffer like Wireshark.
A good example is the UAC with media (uac_pcap) embedded scenario.
SIPp comes with a G711 alaw pre-recorded pcap file and
out of band (RFC 2833) DTMFs in the pcap/ directory.
The PCAP play feature uses pthread_setschedparam calls from pthread library.
Depending on the system settings, you might need to be root to allow this. Please
check "man 3 pthread_setschedparam" man page for details
More details on the possible PCAP play actions can be found in the action reference section.
Exit codes
To ease automation of testing, upon exit (on fatal error or when
the number of asked calls (-m command line option) is
reached, sipp exits with one of the following exit codes:
0: All calls were successful
1: At least one call failed
97: exit on internal command. Calls may have been processed. Also exit on global timeout (see -timeout_global option)
99: Normal exit without calls processed
-1: Fatal error
Depending on the system that SIPp is running on, you can echo
this exit code by using "echo ?" command.
StatisticsResponse times
Response times can be gathered and reported. Response time
names can be arbitrary strings, but for backwards compatibility
the value "true" is treated as if it were named "1". Each
response time can be used to compute time between
two SIPp commands (send, recv or nop). You can start a timer
by using the start_rtd attribute
and stop it using the rtd attribute.
You can view the value of those timers in the SIPp interface
by pressing 3, 6, 7, 8 or 9. You can also save the values
in a CSV file using the -trace_stat option (see below).
If the -trace_rtt option is set, the response times are also dumped
in the >scenario file name<_>pid<_rtt.csv.
Each line represents a RTD measure (triggered by a message reception with a rtd="n" attribute).
The dump frequency is tuned by the -rtt_freq parameter.
Available counters
The -trace_stat option dumps all statistics in
the scenario_name_pid.csv file. The dump starts with one header line
with all counters. All following lines are 'snapshots' of
statistics counter given the statistics report frequency (-fd
option). When SIPp exits, the last values of the statistics
are also dumped in this file.
This file can be easily imported in any spreadsheet
application, like Excel.
In counter names, (P) means 'Periodic' - since last statistic
row and (C) means 'Cumulated' - since sipp was started.
Available statistics are:
StartTime:
Date and time when the test has started.
LastResetTime:
Date and time when periodic counters where last reseted.
CurrentTime:
Date and time of the statistic row.
ElapsedTime:
Elapsed time.
CallRate:
Call rate (calls per seconds).
IncomingCall:
Number of incoming calls.
OutgoingCall:
Number of outgoing calls.
TotalCallCreated:
Number of calls created.
CurrentCall:
Number of calls currently ongoing.
SuccessfulCall:
Number of successful calls.
FailedCall:
Number of failed calls (all reasons).
FailedCannotSendMessage:
Number of failed calls because Sipp cannot send the
message (transport issue).
FailedMaxUDPRetrans:
Number of failed calls because the maximum number of
UDP retransmission attempts has been reached.
FailedUnexpectedMessage:
Number of failed calls because the SIP message received
is not expected in the scenario.
FailedCallRejected:
Number of failed calls because of Sipp internal error.
(a scenario sync command is not recognized or a scenario
action failed or a scenario variable assignment failed).
FailedCmdNotSent:
Number of failed calls because of inter-Sipp
communication error (a scenario sync command failed to
be sent).
FailedRegexpDoesntMatch:
Number of failed calls because of regexp that doesn't
match (there might be several regexp that don't match
during the call but the counter is increased only by
one).
FailedRegexpShouldntMatch:
Number of failed calls because of regexp that shouldn't
match (there might be several regexp that shouldn't match
during the call but the counter is increased only by
one).
FailedRegexpHdrNotFound:
Number of failed calls because of regexp with hdr
option but no matching header found.
FailedOutboundCongestion:
Number of failed outgoing calls because of TCP congestion.
FailedTimeoutOnRecv:
Number of failed calls because of a recv timeout statement.
FailedTimeoutOnSend:
Number of failed calls because of a send timeout statement.
OutOfCallMsgs:
Number of SIP messages received that cannot be associated
with an existing call.
Retransmissions:
Number of SIP messages being retransmitted.
AutoAnswered:
Number of unexpected specific messages received for new Call-ID.
The message has been automatically answered by a 200 OK
Currently, implemented for 'PING' message only.
The counters defined in the scenario are also dumped in the stat file. Counters that have a numeric name are identified by the GenericCounter columns.
In addition, two other statistics are gathered:
ResponseTime (see previous section)
CallLength: this is the time of the duration of an entire call.
Both ResponseTime and CallLength statistics can be tuned using ResponseTimeRepartition
and CallLengthRepartition commands in the scenario.
The standard deviation (STDev) is also available in the log stat file for these two statistics.
Detailed Message Counts
The SIPp screens provide detailed information about the
number of messages sent or recieved, retransmissions, messages lost, and the
number of unexpected messages for each scenario element. Although these
screens can be parsed, it is much simpler to parse a CSV file. To produce a
CSV file that contains the per-message information contained in the main
display screen pass the -trace_counts option. Each column of the file
represents a message and a particular count of interest (e.g., "1_INVITE_Sent"
or "2_100_Unexp"). Each row corresponds to those statistics at a given
statistics reporting interval.
Importing statistics in spreadsheet applicationsExample: importation in Microsoft Excel
Here is a video (Windows Media Player 9 codec or above
required) on how to import CSV statistic files in Excel and
create a graph of failed calls over time.
sipp-02.wmv
Error handling
SIPp has advanced feature to handle errors and unexpected events.
They are detailed in the following sections.
Unexpected messages
When a SIP message that can be
correlated to an existing call (with the
Call-ID: header) but is not expected in the
scenario is received, SIPp will send a CANCEL message if no
200 OK message has been received or a BYE message if a 200
OK message has been received. The call will be marked
as failed. If the unexpected message is a 4XX or 5XX,
SIPp will send an ACK to this message, close the call
and mark the call as failed.
When a SIP message that can't be
correlated to an existing call (with the
Call-ID: header) is received, SIPp will send a
BYE message. The call will not be counted at all.
When a SIP "PING" message is received, SIPp will send
an ACK message in response. This message is not counted as
being an unexpected message. But it is counted in the "AutoAnswered"
statistic counter.
An unexpected message that is not a SIP message will
be simply dropped.
Retransmissions (UDP only)
A retransmission mechanism exists in UDP transport mode.
To activate the retransmission mechanism, the "send" command must include
the "retrans" attribute.
When it is activated and a SIP message is sent and no ACK or
response is received in answer to this message, the message is
re-sent.
The retransmission mechanism follows RFC 3261, section 17.1.1.2.
Retransmissions are differentiated between INVITE and non-INVITE
methods.
<send retrans="500">: will initiate the T1 timer
to 500 milliseconds.
Even if retrans is specified in your scenarios, you can override this by
using the -nr command line option to globally disable the
retransmission mechanism.
Log files
There are several ways to trace what is going on during your SIPp runs.
You can log sent and received SIP messages in <name_of_the_scenario>_<pid>_messages.log by
using the command line parameter -trace_msg. The messages are time-stamped so that you
can track them back.
You also can trace it using the -trace_shortmsg parameter. This logs the most important
values of a message as CSV into one line of the <scenario file name>_<pid>_shortmessages.log
You can trace all unexpected messages or events in <name_of_the_scenario>_<pid>_errors.log by using
the command line parameter -trace_err.
You can trace the counts from the main scenario screen in <name_of_the_scenario>_<pid>_counts.csv by using
the command line parameter -trace_counts.
You can trace the messages and state transitions of
failed calls in <name_of_the_scenario>_<pid>_calldebug.log using
the -trace_calldebug command line parameter. This is useful,
because it has less overhead than -trace_msg yet allows you to
debug call flows that were not completed successfully.
You can save in a file the statistics screens, as displayed in
the interface. This is especially useful when running SIPp in background
mode.
This can be done in different ways:
When SIPp exits to get a final status report (-trace_screen option)
On demand by using USR2 signal (example: kill -SIGUSR2 738)
By pressing 's' key (if -trace_screen option is set)
If the -trace_logs option is set, you can use the <log> action to print some scenario traces in the <scenario file name>_<pid>_logs.log file. See the Log action section
SIPp can treat the messages, short messages, logs, and error logs as ring buffers. This allows you to limit the total amount of space used by these log files and keep only the most recent messages. To set the maximum file size use the -ringbuffer_size option. Once the file exceeds this size (the file size can be exceeded up to the size of a single log message), it is rotated. SIPp can keep several of the most recent files, to specify the number of files to keep use the -ringbuffer_files option. The rotated files have a name of the form <name_of_the_scenario>_<pid>_<logname>_<date>.log, where <date> is the number of seconds since the epoch. If more than one log file is rotated during a one second period, then the date is expressed as <seconds.serial>, where serial is an increasing integer identifier.
Online help (-h)
The online help, available through the -h option is duplicated here for your
convenience
7 calls every 2 seconds.
-r 10 -rp 5s => 10 calls every 5 seconds.
-rate_scale : Control the units for the '+', '-', '*', and '/' keys.
-rate_increase : Specify the rate increase every -fd units (default is
seconds). This allows you to increase the load for each
independent logging period.
Example: -rate_increase 10 -fd 10s
==> increase calls by 10 every 10 seconds.
-rate_max : If -rate_increase is set, then quit after the rate
reaches this value.
Example: -rate_increase 10 -rate_max 100
==> increase calls by 10 until 100 cps is hit.
-no_rate_quit : If -rate_increase is set, do not quit after the rate
reaches -rate_max.
-recv_timeout : Global receive timeout. Default unit is milliseconds. If
the expected message is not received, the call times out
and is aborted.
-send_timeout : Global send timeout. Default unit is milliseconds. If a
message is not sent (due to congestion), the call times
out and is aborted.
-sleep : How long to sleep for at startup. Default unit is
seconds.
-reconnect_close : Should calls be closed on reconnect?
-reconnect_sleep : How long (in milliseconds) to sleep between the close and
reconnect?
-ringbuffer_files: How many error/message files should be kept after
rotation?
-ringbuffer_size : How large should error/message files be before they get
rotated?
-rsa : Set the remote sending address to host:port for sending
the messages.
-rtp_echo : Enable RTP echo. RTP/UDP packets received on port defined
by -mp are echoed to their sender.
RTP/UDP packets coming on this port + 2 are also echoed
to their sender (used for sound and video echo).
-rtt_freq : freq is mandatory. Dump response times every freq calls
in the log file defined by -trace_rtt. Default value is
200.
-s : Set the username part of the resquest URI. Default is
'service'.
-sd : Dumps a default scenario (embeded in the sipp executable)
-sf : Loads an alternate xml scenario file. To learn more
about XML scenario syntax, use the -sd option to dump
embedded scenarios. They contain all the necessary help.
-shortmessage_file: Set the name of the short message log file.
-shortmessage_overwrite: Overwrite the short message log file (default true).
-oocsf : Load out-of-call scenario.
-oocsn : Load out-of-call scenario.
-skip_rlimit : Do not perform rlimit tuning of file descriptor limits.
Default: false.
-slave : 3pcc extended mode: indicates the slave number
-slave_cfg : 3pcc extended mode: indicates the file where the master
and slave addresses are stored
-sn : Use a default scenario (embedded in the sipp executable).
If this option is omitted, the Standard SipStone UAC
scenario is loaded.
Available values in this version:
- 'uac' : Standard SipStone UAC (default).
- 'uas' : Simple UAS responder.
- 'regexp' : Standard SipStone UAC - with regexp and
variables.
- 'branchc' : Branching and conditional branching in
scenarios - client.
- 'branchs' : Branching and conditional branching in
scenarios - server.
Default 3pcc scenarios (see -3pcc option):
- '3pcc-C-A' : Controller A side (must be started after
all other 3pcc scenarios)
- '3pcc-C-B' : Controller B side.
- '3pcc-A' : A side.
- '3pcc-B' : B side.
-stat_delimiter : Set the delimiter for the statistics file
-stf : Set the file name to use to dump statistics
-t : Set the transport mode:
- u1: UDP with one socket (default),
- un: UDP with one socket per call,
- ui: UDP with one socket per IP address The IP
addresses must be defined in the injection file.
- t1: TCP with one socket,
- tn: TCP with one socket per call,
- l1: TLS with one socket,
- ln: TLS with one socket per call,
- s1: SCTP with one socket (default),
- sn: SCTP with one socket per call,
- c1: u1 + compression (only if compression plugin
loaded),
- cn: un + compression (only if compression plugin
loaded). This plugin is not provided with sipp.
-timeout : Global timeout. Default unit is seconds. If this option
is set, SIPp quits after nb units (-timeout 20s quits
after 20 seconds).
-timeout_error : SIPp fails if the global timeout is reached is set
(-timeout option required).
-timer_resol : Set the timer resolution. Default unit is milliseconds.
This option has an impact on timers precision.Small
values allow more precise scheduling but impacts CPU
usage.If the compression is on, the value is set to
50ms. The default value is 10ms.
-T2 : Global T2-timer in milli seconds
-sendbuffer_warn : Produce warnings instead of errors on SendBuffer
failures.
-trace_msg : Displays sent and received SIP messages in __messages.log
-trace_shortmsg : Displays sent and received SIP messages as CSV in
__shortmessages.log
-trace_screen : Dump statistic screens in the
__screens.log file when
quitting SIPp. Useful to get a final status report in
background mode (-bg option).
-trace_err : Trace all unexpected messages in __errors.log.
-trace_calldebug : Dumps debugging information about aborted calls to
__calldebug.log file.
-trace_stat : Dumps all statistics in _.csv file.
Use the '-h stat' option for a detailed description of
the statistics file content.
-trace_counts : Dumps individual message counts in a CSV file.
-trace_rtt : Allow tracing of all response times in __rtt.csv.
-trace_logs : Allow tracing of actions in __logs.log.
-users : Instead of starting calls at a fixed rate, begin 'users'
calls at startup, and keep the number of calls constant.
-watchdog_interval: Set gap between watchdog timer firings. Default is 400.
-watchdog_reset : If the watchdog timer has not fired in more than this
time period, then reset the max triggers counters.
Default is 10 minutes.
-watchdog_minor_threshold: If it has been longer than this period between watchdog
executions count a minor trip. Default is 500.
-watchdog_major_threshold: If it has been longer than this period between watchdog
executions count a major trip. Default is 3000.
-watchdog_major_maxtriggers: How many times the major watchdog timer can be tripped
before the test is terminated. Default is 10.
-watchdog_minor_maxtriggers: How many times the minor watchdog timer can be tripped
before the test is terminated. Default is 120.
-tls_cert : Set the name for TLS Certificate file. Default is
'cacert.pem
-tls_key : Set the name for TLS Private Key file. Default is
'cakey.pem'
-tls_crl : Set the name for Certificate Revocation List file. If not
specified, X509 CRL is not activated.
-3pcc : Launch the tool in 3pcc mode ("Third Party call
control"). The passed ip address is depending on the
3PCC role.
- When the first twin command is 'sendCmd' then this is
the address of the remote twin socket. SIPp will try to
connect to this address:port to send the twin command
(This instance must be started after all other 3PCC
scenarii).
Example: 3PCC-C-A scenario.
- When the first twin command is 'recvCmd' then this is
the address of the local twin socket. SIPp will open
this address:port to listen for twin command.
Example: 3PCC-C-B scenario.
-tdmmap : Generate and handle a table of TDM circuits.
A circuit must be available for the call to be placed.
Format: -tdmmap {0-3}{99}{5-8}{1-31}
-key : keyword value
Set the generic parameter named "keyword" to "value".
-set : variable value
Set the global variable parameter named "variable" to
"value".
-multihome : Set multihome address for SCTP
-heartbeat : Set heartbeat interval in ms for SCTP
-assocmaxret : Set association max retransmit counter for SCTP
-pathmaxret : Set path max retransmit counter for SCTP
-pmtu : Set path MTU for SCTP
-gracefulclose : If true, SCTP association will be closed with SHUTDOWN
(default).
If false, SCTP association will be closed by ABORT.
-dynamicStart : variable value
Set the start offset of dynamic_id varaiable
-dynamicMax : variable value
Set the maximum of dynamic_id variable
-dynamicStep : variable value
Set the increment of dynamic_id variable
Signal handling:
SIPp can be controlled using posix signals. The following signals
are handled:
USR1: Similar to press 'q' keyboard key. It triggers a soft exit
of SIPp. No more new calls are placed and all ongoing calls
are finished before SIPp exits.
Example: kill -SIGUSR1 732
USR2: Triggers a dump of all statistics screens in
__screens.log file. Especially useful
in background mode to know what the current status is.
Example: kill -SIGUSR2 732
Exit code:
Upon exit (on fatal error or when the number of asked calls (-m
option) is reached, sipp exits with one of the following exit
code:
0: All calls were successful
1: At least one call failed
97: exit on internal command. Calls may have been processed
99: Normal exit without calls processed
-1: Fatal error
-2: Fatal error binding a socket
Example:
Run sipp with embedded server (uas) scenario:
./sipp -sn uas
On the same host, run sipp with embedded client (uac) scenario
./sipp -sn uac 127.0.0.1
]]>Performance testing with SIPpAdvices to run performance tests with SIPp
SIPp has been originally designed for SIP performance testing. Reaching
high call rates and/or high number of simultaneous SIP calls is possible
with SIPp, provided that you follow some guidelines:
Use an HP-UX, Linux or other *ix system to reach high performances.
The Windows port of SIPp (through CYGWIN) cannot handle high performances.
Limit the traces to a minimum (usage of -trace_msg, -trace_logs
should be limited to scenario debugging only)
To reach a high number of simultaneous calls in multi-socket mode,
you must increase the number of filedescriptors handled by your system. Check
"Increasing File Descriptors Limit" section for more details.
Understand internal SIPp's scheduling mechanism
and use the -timer_resol, -max_recv_loops and -up_nb command
line parameters to tune SIPp given the system it is running on.
Generally, running performance tests also implies measuring response
times. You can use SIPp's timers (start_rtd, rtd in scenarios and -trace_rtt
command line option) to measure those response times. The precision of those
measures are entirely dependent on the timer_resol parameter
(as described in "SIPp's internal scheduling"
section). You might want to use another "objective" method if
you want to measure those response times with a high precision (a tool
like Wireshark will allow you to do so).
SIPp's internal scheduling
SIPp has a single-threaded event-loop architecture, which allows it to handle
high SIP traffic loads. SIPp's event loop tracks various tasks, most of which are
the calls that are defined in your scenario. In addition to tasks that represent
calls there are several special tasks: a screen update task, a
statistics update task, a call opening task, and a watchdog task. SIPp's main
execution loop consists of:
Waking up tasks that have expired timers.
Running up to max_sched_loop tasks that are in a running
state (each call is executed until it is no longer runnable).
Handling each of the sockets in turn, reading max_recv_loops messages
from the various sockets.
SIPp executes this loop continuously, until some condition tells it
to stop (e.g., the user pressing the 'q' key or the global call limit
or timeout being reached).
Several parameters can be specified on the command line to fine tune
this scheduling.
timer_resol:
during the main loop, the management of calls (management of wait,
retransmission ...) is done for all calls, every "timer_resol" ms at
best. The delay of retransmission must be higher than "timer_resol".
The default timer resolution is 1 millisecond, and that is the most
precise resolution that SIPp currently supports. If you increase
this parameter, SIPp's traffic will be burstier and you are likely
to encounter retransmissions at high load. If you have too many calls,
or each call takes too long, the timer resolution will not be
respected.
max_recv_loops and max_sched_loops:
received messages are read and treated in batch. "max_recv_loops" is the
maximum number of messages that can be read at one time. "max sched loops" is
the maximum number of processing calls loops. These limits prevent
SIPp from reading and processing new messages from sockets to the
exclusion of processing existing calls, and vice versa. For heavy
call rate, increase both values. Be careful, those two parameters
have a large influence on the CPU occupation of SIPp.
watchdog_interval, watchdog_minor_threshold, watchdog_major_threshold,
watchdog_minor_maxtriggers, and watchdog_major_maxtriggers:
The watchdog timer is designed to provide feedback if your call load is
causing SIPp's scheduler to be overwhelmed. The watchdog task sets a timer
that should fire every watchdog_interval milliseconds (by defualt
400ms). If the timer is not serviced for more than
watchdog_minor_threshold milliseconds (by default 500s), then a
"minor" trigger is recorded. If the number of minor triggers is more than
watchdog_minor_maxtriggers; the watchdog task terminates SIPp. Similarly,
if the timer is not serviced for more than watchdog_major_threshold
milliseconds (by default 3000ms), then a major trigger is recorded; and if more
than watchdog_major_maxtriggers are recorded SIPp is terminated. If you only
see occasional messages, your test is likely acceptable, but if these events
are frequent you need to consider using a more powerful machine or
set of machines to run your scenario.
Useful tools aside SIPpJEdit
JEdit (http://www.jedit.org/) is a
GNU GPL text editor written in Java, and available on almost all
platforms. It's extremely powerful and can be used
to edit SIPp scenarios with syntax checking if you put the DTD
(sipp.dtd)
in the same directory as your XML scenario.
Wireshark/tshark
Wireshark (http://www.wireshark.org/) is a
GNU GPL protocol analyzer. It was formerly known as Ethereal. It supports SIP/SDP/RTP.
SIP callflow
When tracing SIP calls, it is very useful to be able
to get a call flow from an wireshark trace. The "callflow" tool allows you to do
that in a graphical way:
http://callflow.sourceforge.net/
An equivalent exist if you want to generate HTML only call flows
http://www.iptel.org/~sipsc/
Getting support
You can likely get email-based support from the sipp users community. The mailing list address is
sipp-users@lists.sourceforge.net.
To protect you from SPAM, this list is restricted (only people that actually subscribed can
post). Also, you can browse the SIPp mailing list archive:
http://lists.sourceforge.net/lists/listinfo/sipp-users
Contributing to SIPp
Of course, we welcome contributions! If you created a feature
for SIPp, please send the "diff" output
(diff -bruN old_sipp_directory new_sipp_directory)
on the SIPp mailing list, so that we can review and possibly integrate it in SIPp.
