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專案描述

Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.

System Requirements

System requirement is not defined
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2006-03-21 05:55 Back to release list
1.1rc5

This release added IPv6 support for the pcapplay feature, AMD64 processors, the ability to reconnect to TCP or TLS connections, enhanced support for strict record routing (tested with SER proxy), and many bugfixes. This release has been tested successfully with up to 1100 simultaneous G711a VoIP calls.
標籤: Unstable, Major feature enhancements

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