Its pleasure to announce the availability of SIPP version 3.1!
Contributors have enriched SIPp with new features, another step towards making a more complete SIP benchmark tool.
A lot of new enhancements and bug fixes have been added since 3.0.
Major changes include-
-Complete rework of the call structure, making the architecture more extensible and enable handling of out-of-call messages.
-Introduction of dead call class, to differentiate a dead call from "out-of-call" messages.... read more
The SIPp contributors are happy to announce you the availability of the new SIPp stable version (3.0) !
A huge number of enhancements and bug fixes have been added since 2.0.1 version was released, including :
- a completely rework of the network I/O subsystem
- a new message parsing procedure
- a more powerful call variables handling, allowing arithmetical operations and string variables... read more
The SIPp contributors are happy to announce the availability of SIPp 2.0!
Since 1.1rc8, many new features and fixes, as well as tons of performance enhancements have been made.
You can get the full release notes there: https://sourceforge.net/project/shownotes.php?release_id=504154
And you can download SIPp 2.0 from: https://sourceforge.net/project/showfiles.php?group_id=104305&package_id=119322&release_id=504154... read more
SIPp 1.1rc8 has just been release.
Since 1.1rc6 (1.1rc7 was never released), many new features and fixes have been made (around 80).
The main one being:
- Performances have been greatly improved by, in some cases, a factor of 10
- Media play is supported on Windows platforms
- Statistical pauses emulate real user behaviors
- TCP/TLS has been fixed on windows platforms,
- Possibility to use SIPp as open loop generator
- Possibility to increase automatically the call rate at specific intervals (benchmarking)
- Fixes for AKA (IMS) authentication... read more
SIPp 1.1rc6 has just been release.
Since 1.1rc5, many new features have been added, the most important one being probably the support of Digest/AKA authentication, provided by Dragos Vingarzan (Fokus) and needed for SIP/IMS scenarios. Not to forget recv_timeouts provided by Peter Higginson. PCAP support has been also greatly enhanced with notably the support of audio+video RTP sending. And... finally... win32 support with authentication (AKA or MD5) is back (without pcap play though).
Full release notes: http://sourceforge.net/project/shownotes.php?release_id=445690&group_id=104305
Since end December 2005, thanks to Guillaume Teissier, SIPp supports replay of pre-recorde pcap audio streams. It allows you, from an Ethereal trace containing both SIP and RTP, to replay what happened.
Since 2006-06-02, SIPp supports also the replay of audio _and_ video streams, making it a great tool not only for SIP but also for audio and video testing.
Julien Gilli from the OpenWengo community speaks about SIPp on the OpenWengo blog (http://blog.openwengo.org/index.php?/archives/2006/03/30.html).
OpenWengo is sponsored by Neuf Cegetel (http://www.groupeneufcegetel.fr/html/en/Home/Home.html), and the community is developing a GPL licensed SIP audio+video softphone and multi-protocol instant messaging client.
Two Open Source projects that have been made to meet!
Here is a new release of SIPp with many additions/bug fixes compared to 1.1rc4.
Thanks to the many of you that participated to this release.
Between RC4 and RC5:
- IPv6 support for pcapplay feature
- Support of AMD64 processors
- Fixed issues when re-using call variables from regexp
- Fixed potential traffic spikes
- Added several changes suggested by Peter Higginson
- Fixed media_ip format in SDP for IPv6 case
- Add a new parameter : max_reconnec: it allows SIPp to try to reconnect after loosing TCP or TLS
- Server can listen different IP address (on the same machine) and client can communicate to
different servers (with different IP address) - Provided by Michel de Boer.
- Fixed a bug where defense messages are sent without using TLS when TLS transport is in use
- Fix issue with RTP threads, all threads were using the same port information
- Remove libnet dependency to build sipp with rtp play support
- Fixed a case with DIGEST authentication failure
- Fixed several issues towards strict record routing - provided by Enrico Hartung.
- Fixed authentication algorithm value that was a quoted-string
- Allow nop operation even if SIPp is not compiled with pcapplay
- Increased max length of CSV files line from 256 to 1024.
- Fixed a problem where SIPp could go in a recusrsive loop
- Added dynamic modification of media port number in IPv4 - in case of reINVITE
SIPp 1.1rc4 brings RTP play capabilities. SIPp is a free Open Source test tool / traffic generator for the SIP protocol. The project is very active and is largely used in the SIP community.
The new pcap play feature (contributed by Guillaume Teissier) allows to send RTP streams and RFC2833 DTMFs along with the SIP traffic. This greatly widen the scope of SIPp and allows SIP compatible equipment provider to test their implementation better than ever. For free.... read more
From RC3 to RC4:
- New pcap play feature (to send RTP streams and RFC2833 DTMFs) - provided by Guillaume Teissier from FT R&D. For Linux only for now. Please read:
- Return of Windows/Cygwin version
- New "nop" command to execute actions only
- Fixed multi-socket mode in UDP that was broken since 2005-02-25
- Fixed bug for offset keyword as well as add negative offset support (+ and - keywords) - provided by Peter Higginson.
- Count unexpected BYE and CANCEL as being unexpected.
- Fixed for bad CSeq in automatic CANCEL (bug 1313890) - Provided by Lee Ballard.
- Fixed default scenarios to accept 180 or 183 (bug 1313881) - Provided by Lee Ballard.
- Fixed rejection of out of cseq method order responses by extending Venkatesh earlier enhancement - reported by Scott Smith.
- Fixed regression compared to sipp 1.0 where call-id headers where not obliged to be the first header - concerns only 3PCC.
- Change in LICENSE.txt to include send_packets.c and send_packets.h specific copyright notice
Here is a new release of SIPp with many additions/bug fixes compared to 1.1rc2. Enjoy :)
- Changes between RC2 and RC3:
* Fixed a type in default regexp scenario
* Added new keywords local_ip_type and media_ip_type so that scenarios can be independent from IPv4/IPv6.
* Changed default scenario with o= local_ip instead of 127.0.0.1.
* Better SSL/TLS error handling with error messages.
* Fixed IPv6 support for media (RTP) ip address.
* Added a warning when buffer size for call variables is not enough.
* Fixed case when SIP entities add Contact without '<', '>' -- this causes SIPp to add bad routes (without '<', '>') - provided by Shriram Natarajan.
* Fix for bug 1279393 (comp.c on *BSD) - provided by Hendrik Scholz.
* New -max_retrans command line parameter to max out UDP retransmissions. This allows to use SIPp to monitor SIP application/servers and detect issues early.
* New [next_url] keyword for a better support of Record-Route - Provided by Shriram Natarajan.
* Support of Certificate Revocation List for TLS transport - needs OpenSSL>=0.9.7. Provided by Venu Bellary.
* Changed Makefile to use ncurses instead of curses - needed by newer Linux OS.
* Fix a crash-on-exit when not using -trace_rtt - provided by Shriram Natarajan.
* Fixed 'addr not supported' being sent as the host when not in IP6 mode - provided by Shriram Natarajan.
* Removed unnecessary getmilliseconds when SIP tracing was not enabled - provided by Bruno Guerin.
* Added 3 command line arguments to fine tune SIPp performances (timer_resol, max_recv_loops, up_nb) - provided by Bruno Guerin.
* Added a check to have at least one mandatory message in a recv sequence.
* Added a check when too many call variables.
* Added binding to one local IP/port which allows using of systems with several IP interfaces - provided by Lord Magnos
* Modified error messages for rsa socket errors
* Added random variable pauses
* bShouldAuthenticate init only when OpenSSL is used - provided by JPeG.
* Fix retransmission of INVITE vs non-INVITE messages per RFC3261 - provided by Venkatesh.
* Do not stop retransmission until a final response is received for non INVITE transactions - provided by Venkatesh.
* Fix stopping retransmission where every response where taken for the request just sent out - provided by Venkatesh.
* Fixed online help to indicate pid instead of ppid - reported by Takahiro Yamashita.
* Catch peer_tag parameter everytime - provided by Nasir Khan.
* Fix init of bShouldAuthenticate boolean which lead to unpredictable behavior during authentication.
* Add display of unexpected messages during pause and display of calls in a pause state - Provided by Peter Higginson.
* Fix bug in stat file name handling and aligned naming rule with other log files.
* Added missing line for TCP congestion fix.
* Fix for outgoing TCP congestion under stress - Provided by Alexandre Ajjan and David Mansutti.
* Fix for max_socket 'cannot get UDP socket' error - Provided by Alexandre Ajjan.
* Fixed handling of a port number different from 5060 on IPv6. Provided by Alexandre Ajjan.
* Fixed using more than 1024 sockets for multi-socket mode. Provided by Alexandre Ajjan.
SIPp 1.1rc2 is available for both Windows and *ix systems.
Many new features and bug fixes.
Here is the changelog:
The 1.1 (Cumulus) branch of SIPp has been introduced in September 2004. The main new features include: IPv6, TLS, Authentication, Conditional branching in scenarios, more actions (log, system exec, abort call), automatic Content-Length computation, sub-expressions in regular expressions.
The latest snapshots are also available as a win32 installer.
Time to move on!
Since 1.0rc3, 1.0final brings the following changes:
- Fix: From F. Tarek Rogers: change all trace file names from pid_scenario to scenario_pid which fixes a pb when using relative paths. Actually enabling -trace_err switch. If -trace_err is not given, unexpected messages are not printed anywhere, except the very last one on exit (along with a message to use -trace_err if desired).
- Fix: fix quitting 3PCC controller with ctrl-C and allow reuse of internal twin socket just after closing.
- Fix: counter in case of receive error that was not incremented
- Fix: bug 917436 (regexp with < or > are not supported). Just need to replace < by < and > by > when specifying regexp.
- Fix: Really disabling retransmission when -nr switch is specified for UDP transport. The fix also benefits to other transports in terms of performances (no more message hash computed).
- Fix: Modified default 3PCC scenario for rtd and response time.
- Fix: bug 1035687 where SDP elements where not properly ordered in default scenarios... read more
1.0rc2 had a compilation issue. Please update to 1.0rc3 or later.
- New: logging for calls that timeout - contributed by Rhys Ulerich.
- New: PID in CSV stat file name
- New: dumping screens in a file on SIGUSR2 signal (check ./sipp -h for usage).
- New: dumping screens on sipp exit (new -trace_screen option).
- New: added process PID in trace files names.
- New: added timestamp in SIP message traces
- New: Make the difference in statistics on unexpected messages for
an existing call (call marked as failed) and unexpected messages for a
non existing call (call not marked as failed, but message counted in new OutOfCallMsgs counter
- New: Added timestamp and call-id in logged error messages
- Fix: compilation issue in 1.0rc2
- Fix: potential memory issues after valgrind tool run
- Fix: wrong CSeq when auto answer on out-of-scenario message.
- Fix: Multiline header CR-LF handing with [last_XXX:] - contributed by Rhys Ulerich.
- Fix: case where a 200 OK has been received and call must be aborted
before ACK is sent (happens in 3PCC).
Very close to 1.0 release, this is 1.0 release candidate 2. Many improvements and some bug fixes in this release compared to 0.3.
Stay tuned for 1.1 which will bring major new features like authentification, TLS and IPV6.
In 2004-06-28 sipp snapshot
(http://sipp.sourceforge.net/snapshots/sipp.2004-06-28.tar.gz), 2 new
features and one bug fix have been added:
1/ Input file feature (contributed by Shriram - thanks!!! - and enhanced
by Joseph). Now you can use "-inf file_name" as a command line parameter
to input values into the scenarios. The first line of the file should
say whether the data is to be read in sequence (SEQUENTIAL) or random
(RANDOM) order. Each line corresponds to one call and has one or more
';' delimited data fields and they can be referred as field0 etc in the
xml scenario file.
Bob;18.104.22.168... read more
The 0.4 release candidate is available for tests.
The main new features compared to 0.3 are:
- Routing recording to support stateful proxies (contributed by Shriram Natarajan)
- Exit code according to call success
- SunOS support (contributed by Shriram Natarajan)
Get it in the un-stable section of sipp: http://sipp.sourceforge.net/snapshots/