Ekiga Manual 4.0

Damien Sandras

Christopher zanee Warner

Matthias Redlich

This documentation is for version 4.0 of Ekiga.

Revision History
Revision Ekiga Manual 4.02012-06-03

Damien Sandras

Revision Ekiga Manual 3.02008-08-31

Damien Sandras

Revision Ekiga Manual 2.02006-01-22

Abstract

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


Table of Contents

Introduction
Ekiga
SIP and H.323
Getting Started
Configuration Assistant Introduction
Personal information
Accounts
Ekiga.net Account
Ekiga Call Out Account
Connection Type
Audio Devices
Video Devices
Configuration Complete
Basic Usage
Calling and being called
Managing Contacts
Sending instant messages
Updating your own status
Managing Calls
Advanced Usage
Registering Additional Accounts
Understanding URIs
Controlling the Video Bandwidth
Monitoring lines
Managing Codecs
Changing Ports
Controlling the SIP and H.323 Settings
About Ekiga
Appendix
Related Software

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 Mac OS X). It was written by Damien Sandras and is licenced 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, etc.. 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 signalling 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.