The SIP Center - A portal for the commercial development of SIP Session Initiation Protocol / Test Area / Testing and Measurement

Testing Tools

Touchstone Technologies provides test and measurement tools and utilities to aid in all aspects of testing SIP voice and video implementations, installations, and services. Ideal for labs, pre-deployment assessments, and post-deployment verification, our products are the most scalable and affordable solutions available from any source. From our free VoIP TraceBuster to our WinSIP load generator capable of delivering perfect media on more than 2,000 concurrent calls, no one can deliver the value-per-dollar proposition of our solutions.

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Extensive SIP call emulation, analysis, monitoring, conformance testing, call trace...

GL's VoIP products PacketGen(TM), PacketScan(TM), RTP Toolbox(TM), Message Automation & Protocol Simulator, and H.323 Call Emulator / Analyzer provide extensive SIP based call emulation, SIP call analysis, SIP call trace, and call monitoring. Fully compliant to current SIP standards and latest codecs, these suite of products provide a comprehensive set of tools for network testing, monitoring, analysis, and evaluation.

Our TDM and Wireless tools complement our VoIP tools to provide an end-to-end simulation and testing environment for Gateways, SIP phones, and ATAs. In addition, GL supports full Voice Quality Testing over TDM, VoIP, and Wireless networks using the industry standard PESQ, PAMS, and PSQM algorithms.

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SIP testing... under GNU GPL license.

Sipp is a GNU GPL open source software, provided to you by Hewlett-Packard engineers. HP does not provide any support or warranty concerning Sipp but you can likely get email-based support from the sipp users community.

Sipp is a performance testing tool for the SIP protocol. It features:

Dependency free compilation (no external library to buy/install)
Works on HPUX, Tru64 and Linux systems
Embedded basic SipStone user agent scenarios (UAC & UAS) that establish and release multiple calls with INVITE and BYE
Accept custom XML scenario files describing SIP call flows
Dynamically adjustable call rate
Dynamic display of statistics about running traffic (active calls, call rate, round trip delay between specified messages, messages count)
Periodic CSV statistics dumps (spreadsheet applications compatible) TCP and UDP (with retransmission management) over one or multiple sockets - enabling a true emulation of thousands of user agents
Posix regular expressions and variables in scenarios - allowing complex call flows

Sipp's homepage >>>

RADVISION's ProLab SIP Test Manager is an advanced testing system capable of performing essential tests including performance, load, stress, interoperability, media and protocol compliance. The ProLab SIP Test Manager is a highly scalable, feature rich testing system that is suitable for use in different stages of the product development cycle. More information >>>

TamoSoft - Software-based real-time SIP and H.323 analysis: call flow; signaling sessions; registrations; media streams; errors; MOS and R-Factor; etc. Available for Ethernet and 802.11 a/b/g/n networks.

More information >>>

Valid8.com - A leading conformance test solutions provider for telecommunications protocols, including VoIP.

The SIP Center TestMessenger allows users to send test messages from text files over UDP to SIP implementations.

Ethereal - Network protocol analyzer

IPC Software tool for displaying SIP call flows

iptel.org SIP Express Router
SIP Express Router (ser) is a high-performance, configurable, free SIP (RFC3261) server . It can act as registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.

PROTOS Test-Suite: c07-sip - The purpose of this test-suite is to evaluate implementation level security and robustness of Session Initiation Protocol (SIP) implementations.

Sipbomber v 0.7
Sipbomber is invaluable tool for SIP developers intended for testing SIP-protocol implementation against rfc3261.
Current version can check only server implementations - (proxies, user agent servers, redirect servers, and registrars).
This program is distributed under terms of GPL.

SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.

sipsak - SIP swiss army knife
sipsak is a small comand line tool for developers and amdinistrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices.

TestYourVoIP.com - A free, voice quality testing portal that enables users to independently measure the quality of their broadband VoIP connections via a quick, self-initiated test call. Version 3.0 lets registered users take advantage of a new "Golden Phone" capability that allows them to call selected TestYourVoIP.com test points from either their mobile, Wi-Fi, traditional, or VoIP phones to measure VoIP quality. The hardware Verifiers at these locations are configured to answer calls just like a real phone and measure the quality of the conversation. This Golden Phone test feature is ideal for those who do not have broadband yet and existing VoIP users who want to see how well their VoIP phones are working. Additionally, visitors to TestYourVoIP.com can now compare the performance of various broadband providers to determine which ones are currently offering the best VoIP quality to their customers.

VOIPSA - VOIPSA's mission is to promote the current state of VoIP security research, VoIP security education and awareness, and free VoIP testing methodologies and tools.

Robustness Testing

IETF SIP Torture Messages
These messages were developed and refined at the SIPIt interoperability test events. During the events problematic messages were noted and released as an IETF-draft. It defines tens of valid and invalid messages, describes them and gives directions as to how the SIP application should react.

PROTOS Test-Suite: c07-sip
The publicly available University of Oulu PROTOS robustness tests contain 4527 test cases in 54 test groups. Varying SIP header fields in INVITE messages are tested. The PROTOS c07-sip test suite uncovered numerous security problems in numerous SIP products that were tested by the developers. This test suite is freely usable, as it is released under the GNU public license.

Codenomicon SIP Test Tool
The Codenomicon SIP Test Tool is the commercial extension of the above work, generated using the Codenomicon test generator. Currently it contains over 20000 test cases in over 2000 test groups and tests all SIP header fields and all SDP fields. It contains a test documentation system and a GUI to assist in the test execution and fault elimination.