There are many things that affect VoIP performance: Packet Loss, Jitter, Packet Delay, latency. All those things are network related and usually most VoIP quality problems are coming from the network. But you are lucky because you have freely available many applications for testing VoIP networks performance in order to achieve a good VoIP deployment.
There are many different tools, most of them are focused on monitoring networks for VoIP but there are also applications for testing SIP security along with performance and many other test.
Wireshark is the world's foremost network protocol analyzer, and is the de facto (and often de jure) standard across many industries and educational institutions.
Wireshark development thrives thanks to the contributions of networking experts across the globe. It is the continuation of a project that started in 1998.
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.
PROTOS test suite:
The Session Initiation Protocol (SIP) is a signalling protocol for Internet telephony, instant messaging and alike. Although SIP implementations have not yet been widely deployed, the product portfolio is expanding rapidly. A subset of SIP, namely INVITE messages, was chosen as the subject protocol for vulnerability assessment through syntax testing and test-suite creation. A survey of the related standards was made. Test-material was prepared and tests were carried out against a sample set of existing implementations. Results were gathered and reported. Many of the implementations available for evaluation failed to perform in a robust manner under the test. Some failures had information security implications, and should be considered as vulnerabilities. In order to achieve a robustness baseline for SIP products this test-material should be adopted for their evaluation and development. A more comprehensive test-suite should be developed as the SIP scene matures.
SIPr pronounced as Sipper is a complete SIP application testing framework ideally suited for feature, interop, regression, acceptance and field simulation.
SIPr is a SIP development framework that makes it easier to develop, deploy and maintain SIP
applications and SIP tests.
The SIP stack in SIPr is the worlds most flexible and configurable SIP stack and that makes it ideally suited to develop test cases on it.
SIP proxy testing tool. Sipbomber is an 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.
Command Line Tools for SIP sessions (complete console based SIP UA) and SIMPLE Presence (Publish, Subscribe, Notify) and XCAP document manipulation
To test SIP SIMPLE client SDK, you can use the Command Line Tools provided by the sipclients package.
Manage global and SIP account settings used by middleware and Command Line Tools
By default the Bonjour account is enabled and set as default. To use the Command Line Tools on the public Internet, you must setup at least a SIP account.
Generates SIP Call Flow diagrams
This is a collection of awk and shell scripts that will take a capture file that can be read by ethereal and produce a callflow sequence diagram. The scripts have been primarily tested with SIP call flows, but should work for other network traffic as well.
The diagrams are nicely done, and you can click on various parts to get additional detail.
SIP testing tool. sipsak is a small comand line tool for developers and administrators of SIP applications. It can be used for some simple tests on SIP applications and devices.
- sends OPTIONS request
- sends text files (which should contain SIP requests)
- traceroute (see section 11 in RFC3261)
- user location test
- flooding test
- random character trashed test
- interpret and react on response
- authentication with qop supported (MD5 and SHA1)
- short notation supported for receiving (not for sending)
- unlimited string replacements in files and requests
- add any header to the requests
- can simulate calls in usrloc mode
- uses symmetric signaling and thus should work behind NAT
- can upload any given contact to a registrar
- send messages to any SIP destination
- Nagios compliant return codes
- search for strings in reply with regluar expression
- use multiple processes to create more server load
- read SIP message from STDIN (e.g. from a pipe '|')
- supports DNS SRV through c-ares or libruli
- supports UDP and TCP transport
miTester for SIP:
SIP testing tool; Automates test execution.
miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
miTester for SIP is the right solution for those people who are work effective and time conscious. The remarkable advantage of miTester for SIP is that we can automate the testing of any SIP Application.
miTester for SIP works on two modes: USER mode and ADVANCED mode of call flows execution.
miTester for SIP architectural model uses the globally accepted SIP Stack and Server programming that offers a range of constructs from simple and very high level to complex-low level interfaces to control and test every aspect of SIP call flows.
Design of this framework includes achieving the test result from simple to complex call flows for all SIP applications. USER and ADVANCED modes of testing is incorporated in this framework which makes the testing process simple.
Simple syntax of Client scripts and server scripts in XML format makes the simulation of call flows easier. Care is taken to cover all test types. miTester for SIP supports RFC standards - RFC 3261, RFC 2976, RFC 3428, RFC 3265, RFC 3262, RFC 3311, RFC 3903, RFC 3455.
miTester for SIP is an open source software project, and is released under the GNU General Public License (GPL). All source code is freely available under the GPL. License
A case study on ADVANCED mode of testing SIP Communicator (V1.0-alpha3-nightly.build.1658) using miTester for SIP is done and published. In this mode, the basic call flows if SIP Communicator is automated. At a click of a button, the test executions complete producing the test reports and test logs.
SIP transaction and call performance measurement tool. The PJSIP.ORG website provides the Open Source, comprehensive, high performance, small footprint multimedia communication libraries written in C language for building embedded/non-embedded VoIP applications.
This sample contains a complete implementation of a SIP performance measurement tool. Unlike other tool such SIPp, pjsip-perf is geared more towards finding the performance of an endpoint by flooding the endpoint with some requests and time the completion of the requests.
SIP Inspector - another good tool on October 12, 2010:
SIP Inspector is yet another tool which is very lean and capable of doing. Can be used to simulate specific SIP signaling scenarios.