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Release 6 notes




History of version 6.xx

Rel 6.06

* Corrected pause duration. Pauses were not that accurate and could be seconds off

* Improved RTP streaming from a pcap file. Bug forced RTP streaming of codec supported. After the

fix it is possible to stream from a file any recorded codec

* Fixed a bug where audio was not locally played if scenario is re-ran multiple times in consecutively

* Fixed a bug which caused outgoing SIP messages not being logged when SIP logging is enabled


Rel 6.04

Bug fixes:

* Fixed RTP echo mode. The bug caused 2 RTP ports being used and possibly 2 RTP packets sent back


Rel 6.02

Bug fixes:

* Fixed the GUI issue which affected users having a lot of local IP addresses.


Rel 6.00

New features:

* SRTP for both live speech or when sending a pcap file

* Ability to play RTP when using WebSockets

* Improved performance to allow high call speeds 300+ cps

* improved logic which allows 300+ simultaneous RTP streams

* when audio is plaid from a pcap file audio progress is kept separately for each remote side

* G729 is supported

* Ability to send multipart SDP bodies

* ability to send hexadecimal content in SDP bodies

* client scenario can have multiple dialogs (registration and outgoing sip calls are now possible in the same scenario file)

* new logic to store licence information to improve the user experience when upgrading an OS, for example

Bug fixes:

* Fixed a bug which caused incorrect logic when switching between playing digits and pauses several times in a row

* RTP streams were not played when switching codec in middle of a call

* sometimes a few packets with incorrect SSRC number were present in an RTP stream

* Numerous other bug fixes

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