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User Testing Platform | Testeum
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The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic...
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed...
VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.
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libfindrtp is a C library providing functions used to identify Real-time Transport Protocol (RTP) session endpoint ports and network addresses from active VoIP signaling traffic.
The purpose of this project is creating a plugin for <a href="http://www.ethereal.com/">Ethereal</a> that will enable it to decode ITU-T Recommendation H.323 traffic. The implementation is based on <a href="http://www.openh323.org">OpenH323</a> code
This is an extension to the project OpenGatekeeper. It enable the routing through the gatekeeper of the RTP (audio and video) and T.120 (data) traffic so no traffic is directly exchanged between communication endpoints.