SIPp 3.6.0

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.

Tags telephony
License GNU GPL
State stable

Recent Releases

3.6.019 Jun 2019 03:15 minor feature: Automatic filenames (trace files, error files, etc..) are now created in the current working directory instead of in the directory of the scenario file. Only validates SSL certficate if CA-file is separately specified!. routes header in UAS scenario's. (, reported by Stefan Mititelu.). 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. Advantages: cleaner code, fewer scenario variables. Drawbacks: possible ICMP port unreachable messages for RCTP and video. Also, no easy way to discern different streams if you want to bombard a single UAS with multiple RTP streams. Add play_dtmf code originally from https://sourceforge.net/p/sipp/patches/50/ (Dmitry Kunilov), then pull #82 and then #141. Compile with pcap-play support, and use it by adding similar to how you use play_pcap_audio. Add RTP payload 96 in your SDP: m=audio media_port RTP/AVP 0 96 97 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtpmap:97 no-op/8000 Exec syntax is where digits can be one or more of "0123456789#*ABCD" and length defaults to 200 and must be between 50 and 2000. Instead of digits a field... keyword is also accepted. Make sure you add enough after play_dtmf. . Add RTP payload 96 in your SDP: m=audio media_port RTP/AVP 0 96 97 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtpmap:97 no-op/8000. Exec syntax is where digits can be one or more of "0123456789#*ABCD" and length defaults to 200 and must be between 50 and 2000. Instead of digits a field... keyword is also accepted. Make sure you add enough after play_dtmf. Add rtp_echo action (pull #259 by Snom Technology). Compile with --with-rtpstream and u
3.6.0_rc122 May 2019 03:15 minor feature: lots of build, mainly with ncurses/curses/tinfo and openssl. Better parsing of Contact header. (.). Proper retrying of media ports during startup. (.).
3.5.214 Jul 2018 07:25 minor feature: lots of build, mainly with ncurses/curses/tinfo and openssl. Better parsing of Contact header. (.). Proper retrying of media ports during startup. (.).
3.5.2_rc103 Jul 2018 06:05 minor feature: Qop-value in authorization Digest. It can only hold a single value (auth, auth-int...) and does not take double quotes, in contrast to. The challenge. Some servers returned a 400 upon receiving this. Compile error on Cygwin.
3.6-dev30 Mar 2016 03:15 minor feature: Qop-value in authorization Digest. It can only hold a single value (auth, auth-int...) and does not take double quotes, in contrast to. The challenge. Some servers returned a 400 upon receiving this. Compile error on Cygwin.
3.5.126 Mar 2016 20:44 minor feature: Fix qop-value in authorization Digest. It can only hold a single value (auth, auth-int, ...) and does not take double quotes, in contrast to the challenge. Some servers returned a 400 upon receiving this. (Issue #191, reported by @artlov.) Fix compile error on Cygwin. (Issue #193, reported by @Gankarloo.)