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Kphone 1.0.2.1
Kphone is a Session Initiation Protocol (SIP) user agent for Linux. more>>
Kphone is a Session Initiation Protocol (SIP) user agent for Linux, which allows Voice over IP (VoIP) connections over the Internet.
Kphone project supports Presence and Instant Messaging, and to some extent video calls between two hosts.
The KphoneSI is based on the KPhone.
The original KPhone was written by Billy Biggs, current version by Pekka Raisio, Jouni Vuorela and Juha Heinanen at Wirlab with suggestions and fixes from the KPhone community around the world.
The KphoneSI was written by Klaus Fleischmann based on the kphone with various contributions from others and is still in devellopment.
This software package is testet on Suse Linux V 9.0, 9.1, 9.2, 9.3.
This software package comes with NO warranty. You should note that by downloading, compling and installing it you agree to full self responsibility.
Read INSTALL for installation instructions, and CHANGES for the latest additions in functionality and COPYING for the General Public License (GPL).
Note: "This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed."
Enhancements:
- I polished up the installation routines (Bug no. 1547721 plus other
- (undocumented) errors)
- new ring app: kphoneringsh, is allows you to use an app of your
- choice for ringing, see the user manual for more.
- The STUN timer now accepts other values than the default value; the
- default is increased to 1200 sec.
- The well known memory leaks are closed now (Bug no. 1532451)
- partners codecs, if supported by KPhone, are now recognized even if
- the matching a=rtpmap line is missing in the SDP-description. (X-Lite
- sends such nasty things).
- Authentication got more protection from proxies running havoc.
- signalling and media transport for UDP are always set to symmetric mode
- bug #1564126,Call Hold is saved
- ALSA will not block any longer, if more than one application tries to
- access it. This is usefull for tests, but do not expect sound!
- Project 64Bit: KPhone SI except SPEEX is made compilable on 64-bit
- compilers (be carefully, there still are runtime errors).
<<lessKphone project supports Presence and Instant Messaging, and to some extent video calls between two hosts.
The KphoneSI is based on the KPhone.
The original KPhone was written by Billy Biggs, current version by Pekka Raisio, Jouni Vuorela and Juha Heinanen at Wirlab with suggestions and fixes from the KPhone community around the world.
The KphoneSI was written by Klaus Fleischmann based on the kphone with various contributions from others and is still in devellopment.
This software package is testet on Suse Linux V 9.0, 9.1, 9.2, 9.3.
This software package comes with NO warranty. You should note that by downloading, compling and installing it you agree to full self responsibility.
Read INSTALL for installation instructions, and CHANGES for the latest additions in functionality and COPYING for the General Public License (GPL).
Note: "This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed."
Enhancements:
- I polished up the installation routines (Bug no. 1547721 plus other
- (undocumented) errors)
- new ring app: kphoneringsh, is allows you to use an app of your
- choice for ringing, see the user manual for more.
- The STUN timer now accepts other values than the default value; the
- default is increased to 1200 sec.
- The well known memory leaks are closed now (Bug no. 1532451)
- partners codecs, if supported by KPhone, are now recognized even if
- the matching a=rtpmap line is missing in the SDP-description. (X-Lite
- sends such nasty things).
- Authentication got more protection from proxies running havoc.
- signalling and media transport for UDP are always set to symmetric mode
- bug #1564126,Call Hold is saved
- ALSA will not block any longer, if more than one application tries to
- access it. This is usefull for tests, but do not expect sound!
- Project 64Bit: KPhone SI except SPEEX is made compilable on 64-bit
- compilers (be carefully, there still are runtime errors).
Download (1.3MB)
Added: 2006-09-28 License: GPL (GNU General Public License) Price:
1126 downloads
Siproxd 0.5.13
Siproxd is a SIP proxy for SIP-based softphones hidden behind an IP masquerading firewall. more>>
Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT).
Siproxd project allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
HOW TO GET STARTED
make sure libosip2 is installed
If your libposip2 libraries are installed in /usr/local/lib, be sure to include this library path to /etc/ld.so.conf
$ ./configure
$ make
$ make install
edit /usr/etc/siproxd.conf according to your situation.
At least configure if_inbound and if_outbound. They must represent the interface names (e.g. on Linux: ppp0, eth1) for the inbound and outbound interface.
edit /usr/etc/siproxd_passwd.cfg if you enable client authentication in siproxd.conf
start siproxd (siproxd does not require root privileges)
$ siproxd
Enhancements:
- Several issues related to 64 bit architectures have been fixed and several minor bugfixes.
<<lessSiproxd project allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
HOW TO GET STARTED
make sure libosip2 is installed
If your libposip2 libraries are installed in /usr/local/lib, be sure to include this library path to /etc/ld.so.conf
$ ./configure
$ make
$ make install
edit /usr/etc/siproxd.conf according to your situation.
At least configure if_inbound and if_outbound. They must represent the interface names (e.g. on Linux: ppp0, eth1) for the inbound and outbound interface.
edit /usr/etc/siproxd_passwd.cfg if you enable client authentication in siproxd.conf
start siproxd (siproxd does not require root privileges)
$ siproxd
Enhancements:
- Several issues related to 64 bit architectures have been fixed and several minor bugfixes.
Download (0.21MB)
Added: 2006-06-20 License: GPL (GNU General Public License) Price:
702 downloads
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