Latest release notes of the VoIP Analyzer Tool
v22.03.14.00
- SipLog2Pcap
- add CONVERSIONS button in GUI
- SipLog2Pcap: convert SIP logs to Wireshark pcap files.
- SIP log conversions supported from following vendors:
- ▪ Anveo
▪ Asterisk
▪ AudioCodes Syslog
▪ AudioCodes SIP ladder diagram files
▪ Broadsoft
▪ Genesys
▪ Lync
▪ Mediatrix Syslog
▪ Mitel
▪ OneAccess (Ekinops)
▪ OpenScape Business (Atos/Unify)
▪ OpenScape 4000 STMI SIP logs (Atos/Unify)
▪ OpenScape Voice RTT traces (Atos/Unify)
▪ Twilio
- ▪ Anveo
- AudioCodes
- convert AudioCodes syslog files to an audit with a filtered AudioCodes syslog file for each unique SIP call [SID=xxx:xxx:xxx]
o Add unique AudioCodes Log as table entry in all the HTML pages after audit
o Support Audit of AudioCodes Debug Recording (ACDR) capture inclusive AudioCodes logs (SIP + RTP + RTCP)
o Convert AudioCodes debug recording capture to a normal Wireshark capture during the Merge functionality (SIP + RTP + RTCP) - Hex2Pcap: convert Wireshark HEX files to Wireshark pcap files
- Tunneled QSIG in SIP message body
o more exact processing of tunneled QSIG in SIP (or SIP-Q): process as HEX
o detect false hits on \r in multipart/mixed body
o Support "Content-Disposition:" as body header for multipart/mixed
- convert AudioCodes syslog files to an audit with a filtered AudioCodes syslog file for each unique SIP call [SID=xxx:xxx:xxx]
- Still show the tool GUI when the license is expired
- Optimize handling of captures on the Null/loopback device
- add VoIPAnalyzerUpdater: check at startup if online any update is available
- Use "frame.number==" in loggings (not WP== anymore)
- Correct gzip issue during merge on Linux
- Anonymize
- o also anonymize optional parameters in the SIP FROM header
o support anonymize of IGMP multicast packets
o do not process DHCP packets during anonymize
- o also anonymize optional parameters in the SIP FROM header
- RTP
- o Show Max RTP delay, max RTP Jitter and Max RTP Skew in RTP HTML table
o correct influence between RTP and DTMF on RTP statistics
- o Show Max RTP delay, max RTP Jitter and Max RTP Skew in RTP HTML table
v22.02.14.00
- correct influence between RTP and DTMF on RTP statistics
- detect IPv6 For Raw captures
- add extra hints in syslog for anonymize
- anonymize ICE candidates in SDP
- SipLog2Pcap: paste DateTime also when year is not available (e.g. for AudioCodes syslog)
- support AVPF and SAVPF in SDP
- clean addition struct memory at the end of an audit/anonymize/merge
- make anonymized DNS names iso mapped IP addresses
- use more natural filename after anonymize (xxx_ANON)
- correct rewriting of SDP length after anonymize
- optimize TCP re-assembly mechanism
- optimize cleanup of leftovers after TCP assembly
- give warning when zip fle is opened
- correct processing of abbreviated headers
- TCP reassembly rework + hexdump in hex2pcap format
- gzip for pcapng
- cleanup/optimize IP fragmentation
- SipLog2Pcap generates incorrect "application/csta+xml" body
- optimize handling of pcapng capture with both ethernet packets as also sll packets
- Only last RTP codec is shown in the HTML RTP overview
- Update logging for abbreviated SIP headers
- Support the SIP NEGOTIATE method
- optimize logging of sip.Call-ID=="xxx"
- Update SipLog2Pcap help file
- SipLog2Pcap: support Mitel Sip log conversion to pcap
- SipLog2Pcap: add Mitel and OpenScape Voice RTT log conversion to pcap
- Correct PCAP with wrong TCP flags (e.g. push flag)
- solve progress bar > 100% issue
- Support detection of IEEE802.3br preemption
- Add proprietary X-RTP-Stat headers
- LinkLayerType as short
- update on IEEE 802.3br detection
- Add proprietar Siemens SIP headers
- Add proprietar ThigSbc SIP "X-" headers
- SipLog2Pcap: add vendor OneAccess for SIP log conversion to pcap
- SipLog2Pcap: also detect abbreviated content-type SIP header "c:"
- Read QSIG in body as bytes, not as chars
- SipLog2Pcap code optimalisation: make separate classes
- support ^M
- SipLog2Pcap: detect splitted SIP headers and combine them
- Add also "SIP Reason header" in HTML table for Register, Options etc
- Show SIP 403 Forbidden in the HTML ERROR tables
- SipLog2Pcap: detect SDP splitted on 2 lines for AudioCodes syslog logs.
- SipLog2Pcap: optimize splitted SIP messages
- SipLog2Pcap: support SIP messages on one line with numerous \n
- SipLog2Pcap: only search for \n if also SIP/2.0 can be found in the string
- add icons to Web projects and put header on index page
- SVG corrupted for "var protmessage0" when \r leftover occurs
- first detection SIP Common Log Format (CLF)
- error in cleanup tcp stream
- siplog2pcap html corrections (header)
- web: correct clear page for Firefox + make buttons
- Show errors/warnings when SIP RFC errors occur
v21.12.27.00
- Support "RAck" SIP header (RFC3262)
- support IP fragmentation with duplicate packets
- adjust handling of fragmented packets
- Adjust default OpenScape SBC ports from 50000-50019 to 50000-50039
- check maximum framesize of 64K after IP or TCP assembly
- do not use duplicate RTP packets for wave files
- cleanup at the end of IP fragments and TCP segments
- hex2pcap for online tool on website https://hex2pcap.voipanalyzertool.com/
- hex2pcap: detect de-chunked data and skip it
- allow a tab character for Hex2Pcap splitting
- allow maximal merged filesize of 4 Gb
- optimize link layer detection
- detect RTP packets when they use well known SIP ports
- allow any SIP response code in the range 100-699
- add remark in HTML when RTP delay is highter than 60ms
- check if the 5 mandatory SIP headers are available during audit
- prevent false hits on RTP and RTCP after SIP BYE
- SipLog2Pcap for online tool on website https://siplog2pcap.voipanalyzertool.com/
- online web tool: make specific page for wrong cases (e.g. wrong input)
- Wireshark OSPF packet not always correct parsed
- support radius and diagram protocol (not in GUI yet)
- adjust parsing of MGCP request line
- adjust parsing of CSeq SIP header
- do not allow SVG drawing with negative x position
- add help file for relevant commands for tshark to text
- culture dependency for DateTime
- use minimal HTML table height when table is empty
- show short names for wave files in HTML tables
- Add SIP reason header in generated HTML tables
- optimize splitting of long strings (e.g. SIP From/To/CallID) in HTML tables
v21.10.02.00
- now supported as Date format: day/month/year or month/day/year or year/month/day
- Support RawIP for Hex to Pcap conversions
- also anonymize SRTP
- give warning for zero duration parameter for DTMF via RFC2833/RFC4733
- use only destination IP and destination port for RTCP filenames
- check if SDP content length value header is consistent with SDP body length
- minor correction for new SDP content length after anonymize
- major performance optimalisation during writing of syslog entries from Pcap to syslog text files
- correction regarding detection of multiple SIP messages in one single Wireshark packet
- support P-RTP-Stat header for RTP statistics
- ignore X-Siemens-RTP-Stats: stats not available
v21.09.16.00
- major performance optimalisation during audit
- - finetune anonymize of MGCP messages
- - minor updates in help files
- - create new CHM help file
- - minor bug fixes
v21.05.12.00
- New code signing certificate
- Detect ZRTP magic cookie in RTP header
- SIP port range extended to 5060-5080 and 5090
- Time processing : Local or UTC (also in GUI)
- Merge folder selection instead of one file selection in the merge folder
- MGCP csv file
- Support nested 802.1Q
- Useragent text added to message flow diagrams
v21.03.31.00
- support of the MGCP protocol
- Handling SVG message graphs more generic (also for MGCP)
- Correction on script error popup for .chm help file
- Correction with multiple linktypes in pcapng
- Filter column buttons in HTML tables
- issue solving
v21.02.22.00
- support Raw IP header
- use colors in RTP HTML pages to highlight issues with packet drop, jitter, skew etc
- support abbreviated SIP headers
- use always TTL value 64 after anonymize to hide network topology
- support detection of GRE tunnel will NULL encryption
- add “clear logging window” button in GUI
- TCP segmentation finetuning
- Detect SIP BYE immediate after SIP RE-INVITE (incomplete SIP dialog)
- Make ethernet address anonymous during anonymize
- Add dropzone on website
- Added free online demo on our https://www.voipanalyzertool.com website
- Optimize performance
v20.11.14.00
- performance optimalisation during merge action
- update of online help files
- update of offline CHM help files
- update tool icon
- bug fixes
v20.10.27.00
- support merging of Ethernet packets + Linux SLL packets at the same time to one wireshark capture
v20.10.26.00
- support the Wireshark snoop format
- use UTC time for Wireshark captures
- make also RTP stats in CSV format
- also anonymize data TCP packets with data length zero (e.g. TCP SYN)
- support VoIPAnalyzer tool on Linux
- support VoIPAnalyzer tool on macOS
- bugfixes
v20.10.04.00
- Support of PCAPNG (pcap next generation) Wireshark format
- Add also a split button next to the merge button
- Improved performance in some area’s
- Add progressbar in GUI
- bugfixes
v20.07.13.00
- Initial version released for any 64 bit Windows Operating system