Sponsored results
Links
- OpenH323 Project
The OpenH323 project aims to create a full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
http://www.openh323.org/
- OpenH323 Gatekeeper
Free H.323 gatekeeper (GPL) based on OpenH323.
http://www.gnugk.org/
- Nautilus Secure Phone
Program that allows two parties to hold a secure voice conversation using Linux and TCP/IP.
http://nautilus.berlios.de/
- Partysip
SIP proxy server. It can operate as registrar server, redirect server and stateful proxy server. SIP is an open standard (IETF) replacement for H323.
http://www.nongnu.org/partysip/partysip.html
- Siproxd Project
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).
http://siproxd.sourceforge.net/
- Vovida
VOCAL is an open source project dealing with VoIP. It includes a SIP based Redirect Server, Feature Server, Provisioning Server and Marshal Proxy.
http://www.vovida.org/
- Skype
Peer to peer voice service. Users may call landlines and cellphones for a fee; users may call each other for free. Source code is not available.
http://www.skype.com/
- SIPSet
Sip based phone. You can use it as a soft phone, to make and receive calls from your Linux PC.
http://www.vovida.org/applications/downloads/sipset/
- NAT H323 under Linux
Mini-HOWTO for NAT & H.323 - configuring Linux to support H.323 while using network address translation.
http://pierre.clerissi.free.fr/uHOWTO/nath323.html
- Minisip
SIP based phone. Supports video, communication encryption and push-to-talk.
http://www.minisip.org/
- GNU Bayonne
A free, scalable telecommunications application server by the GNU Project.
http://www.gnu.org/software/bayonne/bayonne.html
- Gnome-o-Phone
Graphical VoIP software for the Linux Gnome desktop. Includes RtpTunnel utility for tunneling RTP UDP-based traffic through a firewall.
http://gphone.sourceforge.net/
- Ekiga
Videoconferencing application to make audio and video calls to remote users. Supports both SIP and H.323.
http://www.gnomemeeting.org/
- GNU oSIP
Implementation of Session Initiation Protocol (SIP). This library provides an interface to initiate and control SIP based sessions.
http://www.gnu.org/software/osip/osip.html
- isdn2h323
Project homepage for isdn2h323, a Linux-based ISDN to H.323 gateway.
http://www.telos.de/linux/H323/
- Linphone
SIP based web phone. Supported audio codecs are G711, LPC10-15, GSM, and SPEEX.
http://www.linphone.org/
- KPhone
SIP based VoIP phone. It supports Presence and Instant Messaging.
http://www.wirlab.net/kphone/
- Asterisk
Open Source telephony switching and private branch exchange (PBX) daemon. Supported signalling protocols: H.323, SIP, MGCP, SCCP (Cisco Skinny).
http://www.asterisk.org/
The Computer Science Directory currently has categories, which include links along with indexed pages.