Projects / Sipp

Sipp

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.

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Recent releases

  •  27 Apr 2007 15:34

    Release Notes: This release adds statistical conditional branching, a 3PCC extended mode, a tool to monitor remote SIP servers through SNMP, a global commandline option to specify that packets should be lost at a given percentage, and many small enhancements, including major performance improvements. clock_tick is updated more frequently now, for higher timer and statistics resolution (in microseconds).

    •  20 Dec 2006 19:27

      Release Notes: Performance has been greatly improved by a factor of 10 in some cases. Media play is supported on Windows platforms. Statistical pauses emulate real user behavior. TCP/TLS has been fixed on Windows platforms. The ability to use SIPp as an open loop generator has been added. The ability to automatically increase the call rate at specific intervals (benchmarking) has been added. There are over 75 other bugfixes and improvements.

      •  07 Sep 2006 18:14

        Release Notes: This release adds support for Digest/AKA authentication for IMS, timeout on receive, many enhancements for RTP play with support for audio+video RTP streams, the return of the Win32 version, support for SIP/TDM maps, and call_id string customization.

        •  21 Mar 2006 04:55

          Release Notes: 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.

          •  21 Dec 2005 15:51

            Release Notes: A new pcap play feature was added to send RTP streams (voice and RFC2833 DTMFs). A new "nop" command was added to execute actions only. The Windows/Cygwin version has returned. Multi-socket mode in UDP was fixed. Several other bugfixes were made.

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