Ekiga Manual 4.0
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## Ekiga Manual 4.0

### Damien Sandras

### Christopher zanee Warner

### Matthias Redlich

This documentation is for version 4.0 of Ekiga.

Tekijänoikeudet © 2003-2012 Damien Sandras

Tekijänoikeudet © 2003-2004 Matthias Redlich

Tekijänoikeudet © 2003-2004 Christopher Warner

Tekijänoikeudet © 2008. Timo Jyrinki (timo.jyrinki@iki.fi)

Versiohistoria
Versio Ekiga Manual 4.0 | 2012-06-03
Damien Sandras
Versio Ekigan ohjekirja 3.0 | 2008-08-31
Damien Sandras
Versio Ekigan ohjekirja 2.0 | 2006-01-22

Tiivistelmä

Ekiga on monia ääni- ja videokoodekkeja tukeva Internetiä käyttävä sovellus äänen, puheluiden ja videokonferenssien välittämiseen.

## Johdanto

### Ekiga

Ekiga on vapaa Internetin (IP) yli ääntä, puheluita ja videokonferensseja välittävä sovellus Linuxilla ja muille Unixeille (esim. BSD, OpenSolaris ja MacOSX). Sen on kirjoittanut Damien Sandras ja se on lisensoitu GNU/GPL:n alaisuuteen.

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 ja 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.

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