Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.
Elastix is a software appliance that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules. The goals of Elastix are reliability, modularity, and ease-of-use. It also features strong reporting capabilities.
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.
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.
SipUnit provides a class library that allows software developers to create automated unit tests for SIP applications. Session Initiation Protocol (SIP) is widely used for providing telephone services over the Internet. SipUnit extends the JUnit framework to incorporate SIP-specific assertions, and it provides a high-level API for performing the SIP operations needed to interact with or invoke a test target. A test program using the SipUnit API is written in Java and acts as a network element that sends/receives SIP requests and responses.
pcapsipdump is a tool for dumping (recording) SIP sessions (and RTP traffic, if available) to disk in a fashion similar to "tcpdump -w" (the format is exactly the same). The difference is that the data is saved with one file per SIP session. Even if there are thousands of concurrect SIP sessions, each goes to separate file.