Looking for a Windows 7 software that allows you to test and analyze VoIP networks? Look no further than SIPp, developed by esteemed programmer Richard GAYRAUD. With its user-friendly interface and comprehensive features, SIPp makes it easy to simulate SIP endpoints, test call flows, and troubleshoot network issues. Whether you're a seasoned IT professional or a novice user, SIPp has everything you need to optimize your VoIP network performance. Download SIPp now and experience the ultimate VoIP testing tool!

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.

Other advanced features include support of IPv6, TLS, SIP authentication, conditional scenarios, UDP retransmissions, error robustness (call timeout, protocol defense), call specific variable, Posix regular expression to extract and re-inject any protocol fields, custom actions (log, system command exec, call stop) on message receive, field injection from external CSV file to emulate live users.

SIPp can also send media (RTP) traffic through RTP echo and RTP / pcap replay. Media can be audio or audio+video.

While optimized for traffic, stress and performance testing, SIPp can be used to run one single call and exit, providing a passed/failed verdict.

Last, but not least, SIPp has a comprehensive documentation available both in HTML and PDF format.

SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very useful to emulate thousands of user agents calling your SIP system.

New Release
Automatic filenames (trace files, error files, etc..) are now created in the current working directory instead of in the directory of the scenario file. (Issue #399, reported by @sergey-safarov.)
Only validates SSL certficate if CA-file is separately specified! (PR #335, by Patrick Wildt @bluerise.)
Angle brackets < and > need to be escaped inside XML attributes. See #414. So, not regexp=" *<(sip:.*)>" but regexp=" *<(sip:.*)>".
Fix [routes] header in UAS scenario's. (Issue #262, reported by Stefan Mititelu (@smititelu).)
last_Keyword does not search in SIP body anymore (#207, reported by Zoltan).
Added PAGER by default to the extremely large sipp help output.
Removed unused RTPStream code concerning video streams. Also consolidated the rtpstream audio port usage to reuse the global [media_port] instead of the [rtpstream_audio_port]. Also the -min_rtp_port and -max_rtp_port options have been removed.
[ SIPp release history ]

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