Ekiga Manual 4.0
|  | Naprej

## Ekiga Manual 4.0

### Damien Sandras

### Christopher Zanee Warner

### Matthias Redlich

This documentation is for version 4.0 of Ekiga.

Pravna zaščita © 2003-2012 Damien Sandras

Pravna zaščita © 2003-2004 Matthias Redlich

Pravna zaščita © 2003-2004 Christopher Warner

Zgodovina različic
Različica Ekiga Manual 4.0 | 2012-06-03
Damien Sandras
Različica Ekiga Manual 3.0 | 2008-08-31
Damien Sandras
Različica Ekiga Manual 2.0 | 2006-01-22

Povzetek

Ekiga is an application for voice over IP, IP telephony, and video conferencing, with support for many audio and video codecs.

## Introduction

### Ekiga

Ekiga is a free Voice over IP, IP Telephony and Video-Conferencing application for Linux and other Unices (e.g BSD, OpenSolaris or MacOSX). It was written by Damien Sandras and is licensed under the GNU/GPL.

Ekiga is able to use modern Voice over IP protocols like SIP and H.323. It supports all major features defined by those protocols like call hold, call transfer, call forwarding, ... It also supports instant messaging, and presence. It also has advanced support for NAT traversal. Ekiga supports the best free audio and video codecs, and has wideband support for a superior audio quality, together with echo cancellation.

### SIP and H.323

The Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is one of the leading signalling protocols for Voice over IP.

H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs but it has rapidly evolved to address the growing needs of VoIP networks. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks. H.323 is based on the ISDN Q.931 protocol and is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN based PBX systems.

|  | Naprej
|  | Getting Started