Performance Testing with Sipp
With this test enviroment we can simulate a single or multple users calling trought mor/asterisk.
In the scenario uac_auth.xml, sipp start sending a SIP INVITE to MOR and if AUTH is requested, sipp resend SIP INVITE with PROXY AUTHENTICATION, Playing with sipp parameters you can increase call data rate and other variable like call duration, etc.
Download and compile the last sipp:
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz tar xvfz sipp.3.1.src.tar.gz cd sipp.svn apt-get install libssl-dev / yum install openssl-devel make ossl
edit a new file "users.csv":
SEQUENTIAL
user001;[authentication username=user001 password=password001]
user002;[authentication username=user001 password=password001]
user003;[authentication username=user001 password=password001]
edit a new file "uac_auth.xml":
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC with AUTH scenario">
<send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="407" auth="true"> </recv> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 2 INVITE Contact: sip:[field0]@[local_ip]:[local_port] [field1] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- t=0 0 c=IN IP4 [media_ip] m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="183" optional="true"> </recv> <recv response="200" rtd="true"> </recv> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause/> <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true"> </recv> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
Then run this:
./sipp -inf users.csv -sf uac_auth.xml -s <extension_to_call> <mor_sip_server_sip>
For a more complete sipp syntax look here: