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A value of 0 indicates that no payload protocol is specified.

This parameter is a hostname as defined in RFC 1123, section 2.1.

SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, and SIP PBXes.

It is also very useful to emulate thousands of user agents calling your SIP system. Here is a screenshot And here is a video (Windows Media Player 9 codec or above required) of SIPp in action: sipp-01SIPp is released under the GNU GPL license. It was originally created and provided to the SIP community by Hewlett-Packard engineers in hope it can be useful, but HP does not provide any support nor warranty concerning SIPp.

We recommend you test them before activating them in a production environment.

These pages allow you to configure protection against attacks which attempt to target network components or the firewall directly.

So you can be confident that all mentioned features will work :) SIPp allows to generate one or many SIP calls to one remote system. In this example, two SIPp are started in front of each other to demonstrate SIPp capabilities.

Run sipp with embedded server (uas) scenario: Integrated scenarios?

It includes a few basic Sip Stone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods.

If you are encountering a Word Press error message or white screen, don't panic.

Someone will have encountered the same message before and it can easily be solved.

Like many other "open source" projects, there are two versions of SIPp: a stable and unstable release.

Stable release: before being labelled as "stable", a SIPp release is thoroughly tested.

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