Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
gnokii is a multisystem tool suite for mobile phones. It provides a library to communicate with a phone hiding the communication protocol. The library handles SMS, phonebook, calendar, phone calls, and other mobile phone capabilities. It supports Nokia-FBUS mobiles, AT-capable phones (most of the mobiles), as well as Symbian-based phones.
Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.
The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.
Skype uses P2P (peer-to-peer) technology to provide voice- and video-based communication with other Internet users. The technology is extremely advanced, but easy to use. It features excellent sound quality, end-to-end encryption, and automatic negotiation of firewalls or routers. Among major features are SkypeOut and SkypeIn, adding the possibility to make low-cost calls to land line phones and having a fixed number to be able to receive calls from land line phones.
SCMxx is a console program that allows you to exchange certain types of data with mobile phones made by Siemens. Some of the data types that can be exchanged are logos, ring tones, vCalendars, vCards, phonebook entries, and SMS messages. It works with the following phones: S25, C35i, M35i, S35i, ME45, S45, SL45, M50, and probably some others, too. It basically uses the AT command set published by Siemens (with some other additional resources).
SER or SIP Express Router is a very fast and flexible SIP (RFC3261) server. It can handle thousands of calls per second on low-budget hadware. A C shell-like scripting language provides full control over the server's behavior. Its modular architecture allows only required functionality to be loaded. The following modules are available: accounting, digest authentication, CPL scripts, ENUM support, instant messaging, MySQL support, PostgreSQL support, a presence agent, Radius authentication and accounting, Diameter authentication, record routing, an SMS gateway, a Jabber gateway, NAT traversal support transaction module, a registrar, and user location.
Callweaver is a community driven software PBX project. The most important differences between Callweaver and Asterisk are built-in STUN support, the use of SpanDSP for better codecs and full T.38 fax over IP support, Sqlite instead of Berkeley DB, universal jitterbuffer, POSIX timers to avoid Zaptel timing dependencies, greater speed, more efficient dialplan execution, and greater stability.
NetUP UTM is a universal billing system for internet service providers of any size. Its modern approach to traffic accounting makes the system compatible with all popular platforms and network devices. Its key features include realtime traffic processing, Cisco Netflow and IP Accounting data collection, support for RADIUS authentication, and cross-platform compatibility. The core of the system is a smart and reliable accounting engine working directly with network equipment. It supports up to 100,000 users at a total speed of up to 3 Gbps. A flexible ratings engine and efficient administration tools make UTM a complete solution for IP/VoIP/WiFi/dial-up billing.
RAT is an RTP audio conferencing and streaming application that allows users to particpate in point-to-point and multi-point audio conferences over the Internet. Developing features include multiple sampling rates, mono/stereo, loss concealment, and IPv4/IPv6 support, and both versions feature encryption and run on a range of platforms.