SIPp is an open source testing tool and traffic generator for the SIP protocol. It involves main SipStone user agent scenarios (UAC and UAS). It establishes and releases call flow through the INVITE and BYE methods. It is also able to read custom XML scenario files describing performance testing configuration from simple to complex call flows. SIPp is licensed under the GNU General Public License.
It is also good for emulating thousands of user agents who are calling your SIP system.
The dynamic display of statistics on running tests
Periodic CSV statistics dumps
Dynamically adjustable call rates
TCP and UDP over multiple sockets
Regular expressions and variables in scenario files
What is great about SIPp is that it can send RTP media (both audio and video) traffic via RTP echo and RTP/pcap replay. Aside from optimizing traffic, stress and performance testing, SIPp is also used to run one single call and exit, while providing a passed/failed verdict. There is a comprehensive documentation available in SIPp webpage both in HTML and PDF format.
If you need open source testing tool SIPp to generate traffic for the SIP protocol but don’t know how to make the most of this tool. Let us install it properly and help you to increase the performance of your VoIP phone system. IP2VOICE is able to host your software in the cloud providing secure and smooth run of your software. Whether you choose managed or unmanaged hosting, IP2VOICE will give you exceptional service thanks to its powerful and high performance servers.