Um die Verzögerungen auf dem Weg zu messen, bietet sich ein Ping für eine grobe Annäherung an. Dabei sind zwei Faktoren zu beachten: Zum einen misst Ping die Gesamtverzögerung von Hin- und Rückweg (RTT). Da die Sprachdaten aber nur in eine Richtung fließen, muss der Wert halbiert werden. Zum anderen sollten die Testpakete ungefähr so groß sein wie ein Standard-VoIP-Paket.
Geht man von einer Kodierung nach G.711 und 20 ms Sprachdaten pro Paket aus, erhält man eine Nutzlast von 160 Byte (64 Kbit/s * 0,02 s). Dazu kommen noch 40 Byte für die IP/UDP/RTP-Header. Also sollten beim Ping 200 Byte große Pakete verschickt werden. Unter Linux verwenden Sie den Befehl
ping -s 200 <Hostname>
Zum einen erhalten Sie aus dieser Messung einen Anhaltspunkt für die Paketlaufzeit, und zum anderen können Sie damit die Paketverlustrate ermitteln
zum Testen der DL-Geschwindigkeit und Bandbreite, sowie zum Generieren von Last.
http://www.voipmonitor.org/download
VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution.
Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer.
Key features:
* Fast C++ SIP/RTP packet analyzer * Predicts MOS-LQE score according to ITU-T G.107 E-model * Detailed delay/loss statistics stored to MySQL * Each call is saved as standalone pcap file
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.
Installation
apt-get install sip-tester
Doku: sipp-reference.pdf Online: http://sipp.sourceforge.net/doc/reference.html#Main+features
http://www.agnity.com/sipper/
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.
https://www.ee.oulu.fi/research/ouspg/PROTOS_Test-Suite_c07-sip
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.
http://callflow.sourceforge.net/
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.
http://sourceforge.net/projects/mitesterforsip/
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.