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SIPp 3.6.0
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@wdoekes wdoekes released this 18 Jun 20:18
· 330 commits to master since this release
v3.6.0
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wdoekes Walter Doekes
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Learn about vigilant mode.
dadec9a
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wdoekes Walter Doekes
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Learn about vigilant mode.

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, not regexp=" *<(sip:.*)>" but regexp=" *&lt;(sip:.*)&gt;".

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 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 <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/> after play_dtmf.
  • 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 in regress/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
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