Releases: SIPp/sipp
v3.7.3
What's Changed
- Fix various typos by @luzpaz in #679
- More gh actions by @lemenkov in #677
- Update alpine version in dockerfiles by @orgads in #654
- Update gtest to 1.14.0 and bump C++ standard to C++14 by @orgads in #651
- CMake: Increase minimum version to 3.5 by @orgads in #649
- Add define to use local ip hints by @smititelu in #598
- Recover -mp and [auto_media_port] by @orgads in #653
- Don't always reset next rtp port to min port by @smititelu in #635
- Add media support for comfort noise (audio/CN) by @rsvargas in #687
- Allow enabling of TLS verification without supplying CRL file by @IvanRibakov in #663
- Support SHA-256 algorithm (RFC 8760) by @Maratk1n in #676
- Generate random SSRC by @smititelu in #599
- Add
-bind_to_device
option by @igankevich in #630 - Don't block on select() in rtpstream_playrtptask by @acumeeple in #690
- Support TLS 1.3 by @orgads in #695
- Remove support for variable PCAP filenames by @peter-oneill in #698
- Add missing documentation for
variable2
in and by @ticpu in #717 - Added hide and display attributes by @mstovenour in #718
- Fix issue where next_url could return garbage by @mstovenour in #724
- Further explain xml syntax change for 3.6.0 by @mstovenour in #725
- Fix rtp_stream can not bind on mac. and fix \r\n in injection file by @exploman in #729
- Fix rtpstream local port allocation by @viktike in #734
- Resolve full path of pcap which starts with '~' by @rajeshsingh381 in #607
New Contributors
- @luzpaz made their first contribution in #679
- @dependabot made their first contribution in #680
- @rsvargas made their first contribution in #687
- @IvanRibakov made their first contribution in #663
- @Maratk1n made their first contribution in #676
- @igankevich made their first contribution in #630
- @acumeeple made their first contribution in #690
- @peter-oneill made their first contribution in #698
- @ryandesign made their first contribution in #708
- @mstovenour made their first contribution in #718
- @exploman made their first contribution in #729
- @gtjoseph made their first contribution in #731
- @viktike made their first contribution in #734
Full Changelog: v3.7.2...v3.7.3
v3.7.2
v3.7.1
What's Changed
- Control port socket always created (not just for RTP echo) by @yuvalkramer in #636
- Add git to the static build Dockerfiles, so that version.h is generated correctly by @rkday in #625
- Allow build to proceed if gtest is missing by @rkday in #638
New Contributors
- @yuvalkramer made their first contribution in #636
Full Changelog: v3.7.0...v3.7.1
SIPp 3.7.0
NOTE: Please download sipp-3.7.0.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.7.0
tag and lacks version.h
and sipp.1
.
Features added in 3.7.0
- RTPstream can now handle .wav files with a WAV header (by Orgad Shaneh)
Bugs fixed in 3.7.0
- RTPCHECK stability fixes (by Jeannot Langlois)
- Support CRLF-format injection files (by Orgad Shaneh)
- Fix to [next_url] when a display name is present in the contact (by enneig)
- Add 'transport' to the Contact header for UAC scenarios (by Martin Flaska)
- Update built-in scenarios to Copy Record-Route from INVITE to 200OK to comply with RFC 3261 (by kadabusha)
- Fix for local_port keyword using TCP or TLS (by Felippe Silvestre)
- Correct handling of IMS-AKA RES values contianing null bytes (by Sergey Zyrianov)
- Fix potential overwrite of auth value when calculating auth (by ZhaohuiLiu)
- Diagnostics improvements:
- Print, rather than lose, any buffered response time data on exit (by Orgad Shaneh)
- Add the IPs and remote address family to 'Network family mismatch' log (by Rob Day)
- Print OpenSSL error reason when certificate load fails (by Rajesh Singh)
- Give clear error if multiple command-line parameters are being interpreted as remote_host
- Prevent clock_tick moving backwards (and getting behind wheel_base and causing an assert) (by Rob Day)
- Ensure that sockets are marked as non-blocking before OpenSSL calls are made (by Rob Day)
- Prevent RTPStream crash due to a thread ID of 0 (by Rob Day)
- Cygwin, FreeBSD and Hurd build fixes (by Orgad Shaneh, kadabusha and Zopolis4)
- Static build fixes (by Aaron Meriwether)
Checksums
MD5: 7b641ba7eff874b09cc1ccf4f5001547 sipp-3.7.0.tar.gz
SHA1: 3f4ba45cad6245f46354b986d0d859daa23e55b0 sipp-3.7.0.tar.gz
SHA256: 51bd5b58ce258a287ce86fdd027359da8930506b2f61c5e92832c7ed82f643b4 sipp-3.7.0.tar.gz
MD5: 04a339fbe25e9b34af4d71eff96877c0 sipp
SHA1: 2d85f24ad21c25a04883967555004078de6e008f sipp
SHA256: f8c493c54231454a1461e5355072d18c31a6f618950281a41ec1fa996028dc49 sipp
SIPp 3.7.0~rc1
NOTE: Please download sipp-3.7.0~rc1.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.7.0~rc1
tag and lacks version.h
and sipp.1
.
Features added in 3.7.0~rc1
- B2BUA Media Gateway RTP/SRTP bit pattern testing -- see
docs/rtpcheck_xml_syntax_reference.pdf
. Command line examples:By Jeannot Langlois.# UAC (RTP) ./sipp -m 1 -sf sipp_scenarios/pfca_uac_apattern.xml \ -t u1 -i 127.0.0.2 -p 5060 -mp 4000 127.0.0.3:5060 # UAS (RTP) ./sipp -m 1 -sf sipp_scenarios/pfca_uas.xml \ -i 127.0.0.3 -t u1 -p 5060 -mp 5000 -rtp_echo # UAC (SRTP) ./sipp -m 1 -sf sipp_scenarios/pfca_uac_bpattern_crypto_simple.xml \ -t u1 -i 127.0.0.2 -p 5060 -mp 4000 -rtpcheck_debug -srtpcheck_debug \ 127.0.0.3:5060 # UAS (SRTP) ./sipp -m 1 -sf sipp_scenarios/pfca_uas_both_crypto_simple.xml \ -t u1 -i 127.0.0.3 -p 5060 -mp 5000 -srtpcheck_debug
- URL encode/decode
<action>
for scenarios (by Jérôme Poulin). - Variables in the rtpstream/pcap filenames (by Orgad Shaneh).
- WolfSSL/WolfCrypt library support (as alternative to OpenSSL, by Thomas Uhle).
Bugs fixed in 3.7.0~rc1
- Documentation updates. Code cleanups. Build fixes. (By Walter Doekes, Thomas Uhle, ChanderG, Lin Sun, Markus Goetzl, Rob Day, Stefan Mititelu, Orgad Shaneh, Karn Saheb).
- Fix socket/tcp refcount/order issue (by Orgad Shaneh).
- Fix timezone in [date] on FreeBSD (by kadabusha).
- Track auto-answered messages as a visible counter rather than an error log (by Rob Day).
- Unconditionally show index in scenario screen (by Rob Day).
Checksums
MD5: b2a28142424daa6addcf528bb2e9106b sipp-3.7.0~rc1.tar.gz
SHA1: fd825db82e0c099936bfea8d37a24feb812a3d10 sipp-3.7.0~rc1.tar.gz
SHA256: 6195e91117342dc6e8d32b172927a60bf5a1a42a6770ed3d8b31717a7473d2f5 sipp-3.7.0~rc1.tar.gz
SIPp 3.6.2~rc1
NOTE: Please download sipp-3.6.2~rc1.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.6.2~rc1
tag and lacks version.h
and sipp.1
.
Bugs fixed in 3.6.2~rc1
- Fix crash when abusing authentication method (#503, by Markus).
- Fix crash when trying to change an unset ooc scenario (#463, by jquinn60137).
- Fix various build issues with CMake and/or missing version.h and/or compiler warnings. By me, by Silver Chan, Thomas Uhle, Orgad Shaneh.
- Remove RTP_STREAM define. The code is always included. (By Orgad Shaneh.)
NOTE: It is also removed from the version-string! - Various minor documentation fixes. By me, kadabusha, Thomas Uhle, Alexander Traud.
Checksums
MD5: 122319ac100d86d53b0228d4654cff54 sipp-3.6.2~rc1.tar.gz
SHA1: 9d5fcc2ce1fdb49dc3bb4102c93a62074666b91c sipp-3.6.2~rc1.tar.gz
SHA256: 5953b044aa4baaabe93329dbc997abfdffc4b254785fdb0b0d7bbd1436761f6c sipp-3.6.2~rc1.tar.gz
SIPp 3.6.1
NOTE: Please download sipp-3.6.1.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.6.1
tag and lacks version.h
and sipp.1
.
BREAKING(!) changes in 3.6.1
- CMake is now used as build environment: autoconf and friends are gone (#430, by Rob Day (@rkday)). See
build.sh
for CMake invocations. For a full build, do:cmake . -DUSE_GSL=1 -DUSE_PCAP=1 -DUSE_SSL=1 -DUSE_SCTP=1 make -j4
Bugs fixed in 3.6.1
- Consistently unescape XML attributes when loading scenario (#458, by Steve Frécinaux (@nud)).
- Fix buffer overflow in screen output (#479, reported by @brettowe).
- Fix nonce count in auth headers (#421, by Cody Herzog (@codyherzog)).
- Fix parser warning when trying to access 0-byte SDP body (by Lin Sun (@sunlin7)).
- Fix pcapplay on FreeBSD (#434, by Rob Day (@rkday)).
- Improve build validation (#424, by Stanislav Litvinenko (@dolk13)), a few compiler fixes, a few ncurses fixes (including #436, reported by
@TamerL), build cleanup after CMake (#443, #442, by Orgad Shaneh (@orgads)) and libtinfo linker issues (Jeannot Langlois
(@jeannotlanglois)). - Improve provided sipp.dtd file (#425, by David M. Lee (@leedm777)), and XML fixes by Rob Day.
- Make it easier to deal with large SIP packets by adding an optional
CPPFLAGS=-DSIPP_MAX_MSG_SIZE=262144
flag (#422, by Cody Herzog (@codyherzog)).
Checksums
MD5: 89978c2c039385982848f749bafd4cac sipp-3.6.1.tar.gz
SHA1: f8fdf808bdae0d9ab9ab4c405adcc59cf9a30d54 sipp-3.6.1.tar.gz
SHA256: 6a560e83aff982f331ddbcadfb3bd530c5896cd5b757dd6eb682133cc860ecb1 sipp-3.6.1.tar.gz
SIPp 3.6.0
NOTE: Please download sipp-3.6.0.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.6.0
tag and lacks configure files, version.h
and sipp.1
.
BREAKING(!) changes in 3.6.0
- 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, notregexp=" *<(sip:.*)>"
butregexp=" *<(sip:.*)>"
.
Bugs fixed in 3.6.0
- 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).
Changes in 3.6.0
- 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. (Issue #192, reported by @atsakiridis.)
Features added in 3.6.0
- Add
play_dtmf
code originally from https://sourceforge.net/p/sipp/patches/50/ (Dmitry Kunilov), then pull #82 (@horacimacias) and then #141 (@vodik). Compile with pcap-play support, and use it by adding<exec play_dtmf="1234*#"/>
similar to how you useplay_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
<exec play_dtmf="digits[,length]"/>
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
<pause/>
afterplay_dtmf
.
- Add RTP payload 96 in your SDP:
- Add
rtp_echo
action (pull #259 by Snom Technology). Compile with--with-rtpstream
and use it by adding<rtp_echo value="0">
to stop the RTP echo enabled via-rtp_echo
. RTP echo can be restarted via<rtp_echo value="1">
action. Usage example inregress/github-#0259/uas.xml
- Added the required constants for G722 (payload 9) and iLBC at 30ms per frame to rtp_stream media actions. (PR #366, by Jasper Hafkenscheid @hafkensite.)
- Add quick and dirty detection of invalid XML (issue #322).
- Clarify that
-infindex
should takes a basename only (issue #395, reported by @sergey-safarov).
Checksums
MD5: 1fd27333d179d786d3f6a67ee451fae9 sipp-3.6.0.tar.gz
SHA1: 379e5047c45d6f3d35abb3ee57772ebacd8c2b62 sipp-3.6.0.tar.gz
SHA256: e47e7b11fec0769cf76b30623a66390333bdb20323c66043ca535460858fa1bb sipp-3.6.0.tar.gz
SIPp 3.5.3
NOTE: Please download sipp-3.5.3.tar.gz
from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.5.3
tag and lacks configure files, version.h
and sipp.1
.
Bugs fixed in 3.5.3
- Fix
[routes]
header in UAS scenario's. (Issue #262, reported by Stefan Mititelu (@smititelu).) (Backported from b6c7b20 from 3.6.) - Fix bad Content-Length calculation when whitespace was between the CRLF pairs that separate the body. (Issue #337, fixed by Serg Stetsuk (@sergstetsuk)).
- Fix crash in pcap play on send failure because of pthread_cleanup macros. (Issue #74, #370, reported by various people.)
Checksums
MD5: 58dfa2318c1929dd7acb8cb755f87b51 sipp-3.5.3.tar.gz
SHA1: 60c1b214add3ae6c4d30ce0219d05ad0a6c396df sipp-3.5.3.tar.gz
SHA256: 4c411428f511d587907dca59f8dc5763acfcff03fdcee95c548e63bd19e64141 sipp-3.5.3.tar.gz
SIPp 3.6.0~rc1
NOTE: Please download sipp-3.6.0-rc1.tar.gz from link below instead of Source code (tar.gz). They are both source code, but the latter one is a snapshot of the git v3.6.0_rc1
tag and lacks configure files, version.h
and sipp.1
.
Bugs fixed in 3.6.0
- 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).
Changes in 3.6.0
- BEWARE: Only validates SSL certficate if CA-file is separately specified! (PR #335, by Patrick Wildt @bluerise.)
- 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. (Issue #192, reported by @atsakiridis.)
Features added in 3.6.0
- Add
play_dtmf
code originally from https://sourceforge.net/p/sipp/patches/50/ (Dmitry Kunilov), then pull #82 (@horacimacias) and then #141 (@vodik). Compile with pcap-play support, and use it by adding<exec play_dtmf="1234*#"/>
similar to how you useplay_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
<exec play_dtmf="digits[,length]"/>
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
<pause/>
afterplay_dtmf
.
- Add RTP payload 96 in your SDP:
- Add
rtp_echo
action (pull #259 by Snom Technology). Compile with--with-rtpstream
and use it by adding<rtp_echo value="0">
to stop the RTP echo enabled via-rtp_echo
. RTP echo can be restarted via<rtp_echo value="1">
action. Usage example inregress/github-#0259/uas.xml
- Add quick and dirty detection of invalid XML (issue #322).
- Clarify that
-infindex
should takes a basename only (issue #395, reported by @sergey-safarov).