SIPp Introduction

Load and Performance Testing Simon Flannery Ericsson Australia

Agenda
SIPp Basics Running SIPp
Statistics and Screens Command Line Options

Creating Scenarios Authentication Examples

Getting Started
Low-level knowledge of the SIP protocol and SIP signalling inside IMS Scenarios files are written using XML Download from http://sipp.sourceforge.net/ "SIPp is a free Open Source test tool and traffic generator for the SIP protocol "
sipp.sourceforge.net

Used for performance testing and traffic generation for IMT (and can be used for mIMS, but not PoC) Integration & Verification

SIPp Features
Support for both IPv4 and IPv6 RTP media support (PCAP) Call rate distributions
Fixed Uniform Exponential

Support Authentication High Performance and Reliable Allows Complex Sceneries Can be controlled remotely (or externally) Reports Statistics

Running SIPp
SIPp has an interactive and real-time command line interface Quite safe and stable to be left alone, or run in the background (try to always use latest build) Various statistics and reporting screens are available Exit codes:
0 (zero) indicate that all calls were successful 1 indicates that at least one call failed 97 indicates an abnormal exit on internal command 99 indicating a normal exit without processing calls -1 used to indicate a fatal error

The Scenario Screen

Default screen (accessed by pressing 1)

The Statistics Screen

Accessed by pressing 2

Useful Command Line Options

-sf,loads an alternate scenario file -inf, loads an external CSV file to inject values from -r, sets the call rate (calls per second) -rp, specify the rate period (ms) for the call rate (above) Example: -r 15 -rp 1000, 15 calls every second -l, maximum number of simultaneous calls -m, stop and exit when set number of calls are processed

-i, sets the local IP address

-p, sets the local port number

Useful Command Line Options

-auth_uri,forces the domain value of the URI for authentication (for example, 401) -trace_screen,logs the final screens to file (<scenario>_<pid>_screen.log) -trace_err, logs unexpected messages to file (<scenario>_<pid>_errors.log) -trace_stat, logs statistics to file (<scenario>_<pid>.csv) -fd, logs statistics at set interval (seconds) -nd, disable SIPps default behaviors Finally, we end all SIPp commands with the remote host (and port), for example the P-CSCF

SIPp Supports Authentication

Two authentication algorithms are supported:
Digest / MD5 ("algorithm="MD5"") Digest / AKA

When receiving:
401 (Unauthorized) 407 (Proxy Authentication Required)
Simply add auth="true" in the <recv> command to take the challenge into account The authorization header will then be re-injected in the next message using [authentication] keyword

Creating your own Scenarios

Authored using XML notation Unfortunately has spaghetti flow Using labels and jumps (next attributes) Useful elements: <send> <recv> <pause>, speech time <nop>, no operation <label> Registration scenario walk through with authentication Always put optional responses first
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC Basic Registration"> <label id="0" /> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;ex pires=691200 Max-Forwards: 70 Content-Length: 0 ]]> </send> ...to be continued

Creating your own Scenarios

<recv response="200" rtd="true" next="3" optional="true" /> <recv response="403" rtd="true" next="2" optional="true" /> <recv response="404" rtd="true" next="2" optional="true" /> <recv response="401" rtd="true" next="1" auth="true" crlf="true" /> <label id="1" /> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;ex pires=691200 [field2] Max-Forwards: 70 Content-Length: 0 ]]> </send> <recv response="400" rtd="true" next="2" optional="true" /> <recv response="403" rtd="true" next="2" optional="true" /> <recv response="404" rtd="true" next="2" optional="true" /> <recv response="200" rtd="true" next="3" crlf="true" /> <label id="2" /> <nop> <action> <exec int_cmd="stop_call" /> </action> </nop> <label id="3" /> <!-- Definition of the response time repartition table (unit is ms). --> <ResponseTimeRepartition value="30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500" /> <!-- Definition of the call length repartition table (unit is ms). --> <CallLengthRepartition value="30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500" /> </scenario>

Creating your own Scenarios

Keywords from the Command Line
sipp -sf register.xml -inf subscribers_00000_01249.csv -m 1250 -r 200 -rp 1 -l 50000 \ -t u1 -nd -trace_screen -trace_err -i 192.168.22.81 -p 6041 -auth_uri tcm.imt.se \ 192.168.22.58:5060 <scenario name="UAC Basic Registration"> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;expires=691200 Max-Forwards: 70 Content-Length: 0 ]]> </send>

Creating your own Scenarios

Keywords from SIPp
"UDP" or "TCP" Magic cookie (z9hG4bK) + call number + message index Starts from "1" and incremented by 1 for each call

<scenario name="UAC Basic Registration"> <send retrans="500" start_rtd="true"> <![CDATA[

REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;expires=691200 Max-Forwards: 70 Content-Length: 0 ]]> </send>

Automatically generates the CSeq number starting at 1

Identifies a call and is generated by SIPp for each new call (mandatory)

Useful SIPp Keywords

[len], Computed length of the SIP body used in "Content-Length" header [call_number], index of the call starting at 1 and is incremented by 1 for each call [authentication], is used to include the authentication header [msg_index], provides the message number in the scenario [cseq], provides (or generates) the CSeq value of the last (or next) request received

Injecting External Values

Keywords and values from an external file

<scenario name="UAC Basic Registration"> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;expires=691200 Max-Forwards: 70 Content-Length: 0 ]]> </send>

Regular Expression Scenario

<scenario name="UAC Basic Registration"> <label id="0" /> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;expires =60 Max-Forwards: 70 Content-Length: 0 ]]> </send> <recv response="200" rtd="true" next="3" optional="true" /> <!-- Interval Too Brief --> <recv response="423" rtd="true" next="1" crlf="true"> <action> <ereg regexp="[0-9]+" search_in="hdr" header="Min-Expires:" check_it="true" assign_to="1" /> </action> </recv> <label id="1" /> <send retrans="500" start_rtd="true"> <![CDATA[ REGISTER sip:[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number] To: [field0] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact: [field0] <sip:[field0]@[local_ip]:[local_port]>;expires =[$1] Max-Forwards: 70 Content-Length: 0 ]]> </send> <recv response="200" rtd="true" next="3" crlf="true" /> <label id="3" /> <ResponseTimeRepartition value="30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500" /> <CallLengthRepartition value="30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500" /> </scenario>

Definition of a call
Successful call:
Navigates a scenario file from start to finish Does not encounter a stop_call (or other) command

Unsuccessful (failed) call:

Encounters a stop_call, stop_now, stop_gracefully or other?

Example Time
The scenario:
User 46706560000000 calls user 46706560003000, User 46706560000001 calls user 46706560003001,
User 46706560000002 calls user 46706560003002 and so on

Again we will use CSV files to inject who should call who We assume everyone is provisioned and successfully registered We will use two scenario files:
A Caller scenario file (to send INVITEs, pauses and BYEs) A Callee scenario file (to answer the INVITEs and BYEs)

The Caller
CSV Injection File

The Caller

[field0]

[field1]

[field2]

The Callee [field3]

The Caller
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC Basic Invite"> <label id="0" /> <send retrans="500" start_rtd="true"> <![CDATA[ INVITE sip:[field3]@[field1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[field1]>;tag=[call_number]-INVUAC To: [field3] <sip:[field3]@[field1]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: [field0] <sip:[field0]@[local_ip]:[local_port]> Max-Forwards: 70 Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp-UAC 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" rtd="true" /> <recv response="180" rtd="true" /> <recv response="400" rtd="true" next="3" optional="true" /> <recv response="401" rtd="true" next="3" optional="true" /> <recv response="403" rtd="true" next="3" optional="true" /> <recv response="404" rtd="true" next="3" optional="true" /> <recv response="200" rtd="true" next="1" /> <label id="1" /> <send> <![CDATA[ ACK [next_url] SIP/2.0 [last_Via:] [last_From:] [last_To:] [last_Call-ID:] CSeq: [cseq] ACK Max-Forwards: 70 Subject: Performance Test Content-Length: 0 [routes] ]]> </send> <pause milliseconds="60000" next="2" /> optional="true" optional="true" rrs="true" rrs="true" rrs="true" rrs="true" rrs="true"

The Caller
<label id="2" /> <send retrans="500" start_rtd="true"> <![CDATA[ BYE [next_url] SIP/2.0 [last_Via:] [last_From:] [last_To:] [last_Call-ID:] CSeq: [cseq] BYE Max-Forwards: 70 Subject: Performance Test Content-Length: 0 [routes] ]]> </send> <recv response="400" rtd="true" optional="true" /> <recv response="401" rtd="true" optional="true" /> <recv response="403" rtd="true" optional="true" /> <recv response="404" rtd="true" optional="true" /> <recv response="408" rtd="true" optional="true" /> <recv response="200" rtd="true" crlf="true" /> next="3" next="3" next="3" next="3" next="3" next="4" <label id="3" /> <nop> <action> <exec int_cmd="stop_call" /> </action> </nop> <label id="4" /> <!-- 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>

The Caller
Call rate

Local port of the UE

P-CSCF or ACME

The Scenario
Message Counter

Actual speech time

The Callee
CSV Injection File

The Callee [field0] [field1] [field2] No field 3

The Callee
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <send retrans="500"> <![CDATA[

<scenario name="UAS Basic responder">

<recv request="INVITE" rrs="true" crlf="true" /> <send> <![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number]-INV-UAS [last_Call-ID:] [last_CSeq:] Contact: [field0] <sip:[field0]@[local_ip]:[local_port]> [last_Record-Route:] Content-Length: 0 ]]> </send> <pause milliseconds="2000" />

SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number]-INV-UAS [last_Call-ID:] [last_CSeq:] [last_Record-Route:] Contact: [field0] <sip:[field0]@[local_ip]:[local_port]> Content-Type: application/sdp Content-Length: [len]
v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp-UAS 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 request="ACK" optional="true" crlf="true" />

The Callee
<recv request="BYE" /> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: [field0] <sip:[field0]@[local_ip]:[local_port]> Content-Length: 0 ]]> </send> <!-- Keep call open in case the 200 is lost to be able to retransmit it if we receive the BYE again. --> <pause milliseconds="4000" /> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" /> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" /> </scenario>

The Callee only needs to reply to SIP messges from the Caller The Callee has no logic flow The Caller has all the logic flow
Remember all the labels and jumps (next)?

The Callee

If running both the caller and callee on the same computer ensure that they use different ports

The Scenario

References
SIPp documentation http://sipp.sourceforge.net/doc/reference.h tml SIPp in action http://sipp.sourceforge.net/doc/images/sip p-01.wmv Using the Statistics output with Excel http://sipp.sourceforge.net/doc/images/sip p-02.wmv