February 19, 1999 - 3.7 Beta 1
- This release has 2 new features which are the SIP protocol giving
the possibility to invite one or several persons over the
Internet. The SIP implementation is compatible with SDR 2.5.8. The
second feature is the most important but might be restrictive because
when enabled every participant must use Free Phone 3.7b1 ... This is
the Optmal FEC algorithm : You'll find it in the Redundancy Control,
click on Optimal FEC radio button, enter the max bandwith you have
(e.g the maxrate for a modem is 28800, 33600, 56000, for an ISDN line
it's 64000 or 128000 ...) in the MaxRate input box and let's Free
Phone takes care of the rest. Indeed, Free Phone will find the best
way to minimize the impact of losses between you and the other
NB: Actually, this algorithm works only for unicast
December 22, 1997 - 3.5 Beta 3
- This release of Free Phone might not look very different from R3.2B1, but
the internal structure of the code has been completely changed.
We removed old C code and replaced it by C++ classes to facilitate
the evolution of the code in the future. We also fixed assorted
bugs (some might still be lurking in there...).
This version (under Windows with DirectX5.0) supports the 3D Sound and
is able to run with several audio cards in a same PC.
April 1, 1997 - 3.2 Beta 1
- This release of Free Phone supports stereo and the L16 codec. Free
Phone internally sets the hardware to the best configuration (i.e. 16 bits,
stereo, 48 KHz) and then do under/over-sampling and mono/stereo conversions
to send the packets as requested (e.g. mono, 8 KHz) and converts what is
received to the internal configuration. Free Phone now frees the audio
devices (rec/play) while idle.
December 17, 1996 - 3.0 Beta 1
- This release contains a hack to allow the distributed game MiMaze
to stablish a connection with Free Phone which accepts it without
any negotiation. This is just to allow the game talk with Free
Phone up to the time some better mechanism is defined and implemented.
November 25, 1996 - 3.0 Beta 0
- This release supports the Internet draft draft-perkins-rtp-redundancy-04.txt
and hence achieves compatibility with RAT 2.6a14 (or later). That means
this release is not backwards compatible with previous releases of Free
Phone which encapsulate redundancy with a different scheme. In the absence
of redundancy or high quality audio, all releases of Free Phone are compatible.
Internet drafts can be retrived from the shadow directories on ftp.is.co.za
(Africa), nic.nordu.net (Europe), munnari.oz.au (Pacific Rim), ds.internic.net
(US East Coast), or ftp.isi.edu (US West Coast).
October 24, 1996 - 2.0 Beta 6
- You are now able by double clicking on a participant to obtain the
raw loss rate and loss rate after redundancy reconstruction.
October 18, 1996 - 2.0 Beta 5
- A patch has been added to fix a problem of interaction between Free
Phone and the audio driver of Sparc-Ultra/Solaris , if you already
have 2.0b4 version and you are not going to use an Sparc-Ultra
, you don't need to download the 2.0b5 version (of course, a
general solution will (must) be implemented).
September 19, 1996 - 2.0 Beta 4
- A stand alone version has been made available using the ITcl
- A bug where the lost rate field of the RTCP reports used to be flooded
fooling the auomatic control algorithm has been fixed.
- Some improvement to the interface have been made.
Jun 20, 1996 - 2.0 Beta 2
- Changed to Tcl 7.5 and Tk 4.1.
- The GUI has been improved with vu meters and other enhancements.
- A new scheme to manage redundancy and high quality audio has been implemented
using the special RTP payload types
37 and 38. A consequence of this is that versions 2.0 b2 and above
are not compatible with earlier versions (that relied on the extension
bit to encode redundant information).
- The high quality audio is now transparent to users, i.e. participants
can use different sampling rates and yet they can talk to each other.
May 2, 1996 - 1.1 Beta 4
- The address specification was extended in order to support the TTL
specification in a by address basis.
For exemple, you can now specify:
- 220.127.116.11/64 => port=5004/5005, TTL=64
- 18.104.22.168/6000/32 => port=6000/6001, TTL=32
A TTL is interpreted if you have only two arguments (separarated by
`/') and the second one is an integer less than or equal to 255. If 3 arguments
are specified, the second one is the port specification, and the third
one is taken as the TTL.
Previously we only were able to specify a TTL in the command line, and
it was taken for all the multicast groups you joined.
April 1996 - 1.1 Beta 3
First public release of Free Phone.
May 1995 - 0.8
The redundancy mechanism has been implemented using the X bit in RTP.
First tests of a feedback mechanism that adjusts the amount of redundant
information based on RTCP loss reports (see report