Find some brief explanation of the most important technical terms and protocols mentioned in this document:
VoIP stands for voice over Internet protocol. This technology allows the transmission of ordinary telephone calls over the Internet using packet-linked routes. The voice information is sent in discrete packets like any other data transmitted over the Internet via IP.
SIP stands for session initiation protocol. This protocol is used to establish media sessions over networks. In a Linphone context, SIP is the magic that triggers the ring at your counterpart's machine, starts the call, and also terminates it as soon as one of the partners decides to hang up. The actual transmission of voice data is handled by RTP.
RTP stands for real-time transport protocol. It allows the transport of media streams over networks and works over UDP. The data is transmitted by means of discrete packets that are numbered and carry a time stamp to allow correct sequencing and the detection of lost packages.
A DTMF encoder, like a regular telephone, uses pairs of tones to represent the various keys. Each key is associated with a unique combination of one high and one low tone. A decoder then translates these touch-tone combinations back into numbers. Linphone supports DTMF signalling to trigger remote actions, such as checking voice mail.
Codecs are algorithms specially designed to compress audio and video data.
Jitter is the variance of latency (delay) in a connection. Audio devices or connection-oriented systems, like ISDN or PSTN, need a continuous stream of data. To compensate for this, VoIP terminals and gateways implement a jitter buffer that collect the packets before relaying them onto their audio devices or connection-oriented lines (like ISDN). Increasing the size of the jitter buffer decreases the likelihood of data being missed, but the latency of the connection is increased.