SIPp on Linux to Generate Voice Load


This lab introduces the installation and basic use of SIPp, a free voice performance testing tool. It can emulate thousands of user agents making SIP calls and generate both SIP and RTP traffic. Although it is not as powerful as Cyara, contact centre testing tool (, it is FREE, so good for lab or limited budget.

Please visit SIPp website for product details:

Install SIPp on Ubuntu

Step 1: Install Ubuntu server

Please refer to Python to Manage Network Devices (1): Linux lab setup

Step 2: Download and Save SIPp on Ubuntu

SIPp 3.3, current stable version, is available from:


Save ‘sipp-3.3.tar.gz’ on ubuntu server. I downloaded the sipp-3.3.tar.gz from the lab ftp server and saved under the user’s home directory ‘/home/jsmith’.

jsmith@ubuntu:~$ ftp 192.168.x.x
Connected to 192.168.x.x.
220 Microsoft FTP Service
Name (192.168.x.x:jsmith): axxxx
331 Password required for axxxx.
230 User logged in.
Remote system type is Windows_NT.
ftp> bin #use binary transfer instead of ascii to transfer file as raw data.
200 Type set to I.
ftp> dir
200 PORT command successful.
125 Data connection already open; Transfer starting.
ftp> get sipp-3.3.tar.gz
ftp> bye
221 Goodbye.
jsmith@ubuntu:~$ ls #verify the file existence on ubuntu.

Step 3: Compile SIPp on Ubuntu

3. 1 Pre-requisites to compile SIPp are as below:

sudo apt-get install build-essential
sudo apt-get update
sudo apt-get install ncurses-dev
sudo apt-get install g++
sudo apt-get install libpcap0.8-dev
sudo apt-get install libcap-dev

3.2 Execute command  sudo tar -xvzf sipp-3.3.tar.gz to unzip ‘sipp-3.3.tar.gz’ to ‘sipp-3.3’. ‘z’ attribute is to unzip ‘gz’.

3.3  The folder ‘sipp-3.3’ contains all the extracted  files and at the same location as where ‘sipp-3.3.tar.gz’ stays. Execute cd sipp-3.3 to go into the folder.


3.4 Execute ‘sudo make pcapplay’ to compile with pcap files to provide RTP traffic.


Step 4: Generate SIP and RTP Traffic

Deploy two SIPp servers, one with UAC (user agent client, send SIP requests) role and the other with UAS (user agent server, receive SIP request) role, following steps in Step 3.

Execute the following commands under ‘sip-3.3’ folder.


sudo ./sipp -sn uac_pcap 
#'-sn' means using sipp executable built-in scenario.Use '-sf' if self-defined test scenario (xml file) is required.
# '' is the UAS IP.
sudo ./sipp -sn uac -r 7 -rp 2000
# '-r' call rate, how many calls per second
# '-rp' rate period for the call rate


sudo ./sipp -sn uas

Use ./sipp --help or ./sipp | grep xx to check command syntax as below:



After running sudo ./sipp -sn uac_pcap on UAC and sudo ./sipp -sn uas on UAS, the following monitoring windows appear:

TCPdump can be used to capture traffic as below:

admin@VOICE01:~/sipp-3.3$ sudo tcpdump -n -s 0 -v -c 10 port 5060
[sudo] password for admin: 
tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
16:43:55.951552 IP (tos 0x0, ttl 60, id 975, offset 0, flags [DF], proto UDP (17), length 558) > SIP, length: 530
        INVITE sip:service@ SIP/2.0
        Via: SIP/2.0/UDP;branch=z9hG4bK-1334-778-0
        From: sipp <sip:sipp@>;tag=1334SIPpTag00778
        To: service <sip:service@>
        Call-ID: 778-1334@
        CSeq: 1 INVITE
        Contact: sip:sipp@
        Max-Forwards: 70
        Subject: Performance Test
        Content-Type: application/sdp
        Content-Length:   133

        o=user1 53655765 2353687637 IN IP4
        c=IN IP4
        t=0 0
        m=audio 6000 RTP/AVP 0
        a=rtpmap:0 PCMU/8000

Prime Infrastructure is used to monitor the application traffic and shows sip and rtp traffic as below:



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