SIPp — SIP Connectivity Testing
SIPp is an open-source command-line tool that simulates SIP agents (UAC/UAS) and tests the stability of your SIP trunks.
Installation
apt-get install sipp # Debian/Ubuntu yum install sipp # CentOS/RHEL
Main options
| Option | Description | Example |
|---|---|---|
-r <rate> | Call generation rate (calls/sec) | -r 10 |
-l <limit> | Maximum simultaneous calls | -l 100 |
-m <count> | Total number of calls before stopping | -m 500 |
-sf <file> | Custom XML scenario | -sf uac.xml |
-s <service> | Called number/service | -s 0033XXXXXXXXX |
Standard UAC test (outbound calls)
sipp -sn uac 192.168.1.100 -r 5 -l 50 -m 200 -s 0612345678
This test generates 5 calls per second to the SIP server 192.168.1.100, with a maximum of 50 simultaneous calls, for a total of 200 calls.
Stepped test (stress test)
Gradually increase the load to identify the breaking point:
sipp -sn uac sip.webtic.online -r 1 -l 10 -m 50 sipp -sn uac sip.webtic.online -r 5 -l 50 -m 200 sipp -sn uac sip.webtic.online -r 10 -l 100 -m 500 sipp -sn uac sip.webtic.online -r 20 -l 200 -m 1000
Key metrics to monitor
- Call Completion Rate: % of calls receiving a 200 OK response (target > 95%)
- Response Time: INVITE → 200 OK delay (target < 1 second)
- Failed calls: 408 Timeout, 503 Service Unavailable → network problem or overload
Interpreting the results
SIPp displays statistics in real time. At the end of the test, a summary CSV file is generated. A success rate < 90% on your WebTic trunk indicates a configuration or capacity problem.
Support: Contact our team with the SIPp CSV file for a quick analysis.
