SIPp Tool:
SIPp is a stress or performance test tool / traffic generator for the SIP protocol. It can work in both Scenarios (UAC /UAS) and establishes and releases multiple calls with the INVITE and BYE methods.It can also reads Custom XML scenario files describing from very simple to complex call flows. 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 and dynamically adjustable call rates.
SIPp can also send media (RTP) traffic through RTP echo and RTP /pcap replay. Media can be audio or video.
SIPp can be used to test various real SIP equipment 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.
Installing SIPp on linux(Ubuntu):
Installing SIPP on
Ubuntu 12.4 linux
- Run Terminal on Ubuntu
- Goto root using su - and Pwd
- apt-get or Yum intsall sipp-XXX-tar.gz
- cd Sipp-XX
- ls -l
- apt-get install openssl
- apt-get install libssl-dev
- apt-get install libssl1.0.0
- apt-get install libssl0.9.8
- apt-get install pcaputils
- apt-get install libssl-dev
- apt-get install libncursesw5-dev
- apt-get install libncurses5-dev
- apt-get install libcap-dev
- ./sipp XXX.XX.XX.XXX -sf REGISTER_client.xml -inf Register_client.csv -m 1
If you get any error as below
tring to install sipp on ubuntu I get this error
qwerty at qwerty-desktop:~/sipp.2007-02-08$ sudo make
make OSNAME=`uname|sed -e "s/CYGWIN.*/CYGWIN/"`
MODELNAME=`uname -m|sed "s/Power Macintosh/ppc/"` sipp
make[1]: Entering directory `/home/qwerty/sipp.2007-02-08'
gcc -D__LINUX
-pthread -D__3PCC__ -I.
-I/opt/openssl/include -c -o scenario.o
scenario.cpp
gcc: error trying to exec 'cc1plus': execvp: No such file
or directory
make[1]: *** [message.o] Error 127
make[1]: Leaving directory `/home/qwerty/sipp.2007-02-08'
make: *** [all] Error 2
use these commands
sudo
apt-get install -f
sudo apt-get remove libncurses5-dev libncurses5 libtinfo5 ncurses-bin
sudo apt-get update
sudo apt-get upgrade
sudo apt-get install build-essential linux-image linux-image-generic libncurses5-dev libncurses5
SIPP COMMANDS:
sipp remote_host[:remote_port] [options]
In place of Options use the bellow commands required
SIPp Scripts:
Register UAC:
Register.csv :
Register.csv file should contain the User data in below format
SEQUENTIAL
1000;10.XXX.XXX.XX;[authentication username=1000 password=123456];
Register.xml:
Register.xml file should contain below messages
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="register_client">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
[authentication username=21613 password=123456]
Max-Forwards: 70
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="200" rtd="true">
</recv>
</scenario>
INVITE UAC :
Invite.CSV:
SEQUENTIAL
1000;10.XXX.XXX.XXX;[authentication username=1000 password=123456];1001;
INVITE.XML
Invite.xml should contain below message format
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
<scenario name="UAC INvite ">
<send retrans="500" start_txn="invite">
<![CDATA[
INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[field3]@[field1]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
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" response_txn="invite">
</recv>
<recv response="180" optional="true" response_txn="invite">
</recv>
<recv response="200" rtd="true" rrs="true" response_txn="invite">
</recv>
<send ack_txn="invite">
<![CDATA[
ACK sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
</send>
<pause milliseconds="10000"/>
<send retrans="500">
<![CDATA[
BYE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
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>
sudo apt-get remove libncurses5-dev libncurses5 libtinfo5 ncurses-bin
sudo apt-get update
sudo apt-get upgrade
sudo apt-get install build-essential linux-image linux-image-generic libncurses5-dev libncurses5
SIPP COMMANDS:
sipp remote_host[:remote_port] [options]
In place of Options use the bellow commands required
-v : Display version and copyright information.
-aa : Enable automatic 200 OK answer for INFO and NOTIFY
messages
-bind_local : Bind socket to local IP address, i.e. the local IP
address is used as the source IP address. If SIPp runs
in server mode it will only listen on the local IP
address instead of all IP addresses.
-i : Set the local IP address for 'Contact:','Via:', and
'From:' headers. Default is primary host IP address.
-l : Set the maximum number of simultaneous calls. Once this
limit is reached, traffic is decreased until the number
of open calls goes down. Default:
(3 * call_duration (s) * rate).
-m : Stop the test and exit when 'calls' calls are processed
-mi : Set the local media IP address
-max_recv_loops : Set the maximum number of messages received read per
cycle. Increase this value for high traffic level. The
default value is 1000.
-max_reconnect : Set the the maximum number of reconnection.
-max_retrans : Maximum number of UDP retransmissions before call ends on
timeout. Default is 5 for INVITE transactions and 7 for
others.
-max_invite_retrans: Maximum number of UDP retransmissions for invite
transactions before call ends on timeout.
-max_non_invite_retrans: Maximum number of UDP retransmissions for non-invite
transactions before call ends on timeout.
-max_socket : Set the max number of sockets to open simultaneously.
This option is significant if you use one socket per
call. Once this limit is reached, traffic is distributed
over the sockets already opened. Default value is 50000
-mb : Set the RTP echo buffer size (default: 2048).
-mp : Set the local RTP echo port number. Default is 6000.
-nd : No Default. Disable all default behavior of SIPp which
are the following:
- On UDP retransmission timeout, abort the call by
sending a BYE or a CANCEL
- On receive timeout with no ontimeout attribute, abort
the call by sending a BYE or a CANCEL
- On unexpected BYE send a 200 OK and close the call
- On unexpected CANCEL send a 200 OK and close the call
- On unexpected PING send a 200 OK and continue the call
- On any other unexpected message, abort the call by
sending a BYE or a CANCEL
-nr : Disable retransmission in UDP mode.
-p : Set the local port number. Default is a random free port
chosen by the system.
-pause_msg_ign : Ignore the messages received during a pause defined in
the scenario
-r : Set the call rate (in calls per seconds). This value can
bechanged during test by pressing '+','_','*' or '/'.
Default is 10.
pressing '+' key to increase call rate by 1,
pressing '-' key to decrease call rate by 1,
pressing '*' key to increase call rate by 10,
pressing '/' key to decrease call rate by 10.
If the -rp option is used, the call rate is calculated
with the period in ms given by the user.
-rp : Specify the rate period in milliseconds for the call
rate. Default is 1 second. This allows you to have n
calls every m milliseconds (by using -r n -rp m).
Example: -r 7 -rp 2000 ==> 7 calls every 2 seconds.
-rate_increase : Specify the rate increase every -fd seconds. This allows
you to increase the load for each independent logging
period.
Example: -rate_increase 10 -fd 10
==> 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 -max_rate 100
==> increase calls by 10 until 100 cps is hit.
-recv_timeout : Global receive timeout in milliseconds. If the expected
message is not received, the call times out and is
aborted.
-reconnect_close : Should calls be closed on reconnect?
-reconnect_sleep : How long to sleep between the close and reconnect?
-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.
-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,
- c1: u1 + compression (only if compression plugin
loaded),
- cn: un + compression (only if compression plugin
loaded).
-timeout : Global timeout in seconds. If this option is set, SIPp
quits after nb seconds.
-timer_resol : Set the timer resolution in 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.
-trace_msg : Displays sent and received SIP messages in <scenario file
name>_<pid>_messages.log
-trace_screen : Dump statistic screens in the
<scenario_name>_<pid>_ s.log file when quitting
SIPp. Useful to get a final status report in background
mode (-bg option).
-trace_err : Trace all unexpected messages in <scenario file
name>_<pid>_errors.log.
-trace_timeout : Displays call ids for calls with timeouts in <scenario
file name>_<pid>_timeout.log
-trace_stat : Dumps all statistics in <scenario_name>_<pid>.csv file.
Use the '-h stat' option for a detailed description of
the statistics file content.
-trace_rtt : Allow tracing of all response times in <scenario file
name>_<pid>_rtt.csv.
-trace_logs : Allow tracing of <log> actions in <scenario file
name>_<pid>_logs.log.
-up_nb : Set the number of updates of the internal clock during
the reading of received messages. Default value is 1.
-ap : Set the password for authentication challenges. Default
is 'password
-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 : key value
Set the generic parameter named "key" to "value".
SIPp Scripts:
Register UAC:
Register.csv :
Register.csv file should contain the User data in below format
SEQUENTIAL
1000;10.XXX.XXX.XX;[authentication username=1000 password=123456];
Register.xml:
Register.xml file should contain below messages
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="register_client">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
[authentication username=21613 password=123456]
Max-Forwards: 70
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="200" rtd="true">
</recv>
</scenario>
INVITE UAC :
Invite.CSV:
SEQUENTIAL
1000;10.XXX.XXX.XXX;[authentication username=1000 password=123456];1001;
INVITE.XML
Invite.xml should contain below message format
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
<scenario name="UAC INvite ">
<send retrans="500" start_txn="invite">
<![CDATA[
INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[field3]@[field1]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
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" response_txn="invite">
</recv>
<recv response="180" optional="true" response_txn="invite">
</recv>
<recv response="200" rtd="true" rrs="true" response_txn="invite">
</recv>
<send ack_txn="invite">
<![CDATA[
ACK sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
</send>
<pause milliseconds="10000"/>
<send retrans="500">
<![CDATA[
BYE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
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>
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