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2011-09-11T08:42:36Z

Created page with "<big>'''Analysis of VoIP'''</big> == VoIP software == * Asterisk - A free PBX-system * Trixbox - A Linux-distro specially configured for Asterisk ** Trixbox 1.2.3 - O..."

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<big>'''Analysis of VoIP'''</big>
== VoIP software ==
* [[Asterisk]] - A free PBX-system
* [[Trixbox]] - A Linux-distro specially configured for Asterisk
** [[Trixbox 1.2.3]] - Older version
** [[List of IP's of our trixboxes]]

* [[Used software]] - Full list of the software we used.
** [[X-Lite]] - Free SIP soft-phone

* [http://en.wikipedia.org/wiki/List_of_Asterisk_PBX_distributions List of Asterisk PBX distributions]

== Asterisk Configuration ==
* [[Dialplan Basics]] - The dialplan defines how Asterisk handles inbound and outbound calls.
** [[Pattern-matching syntax]] - Patterns used to match a dialed number, for routing purposes in a dialplan.
* [[Videoconferencing]] - Configuration to make a videocall
* [[Screenshots VoIPBuster-config]] - Screenshots describing the configuration of VoIPBuster in Asterisk.
* [[DUNDi configuration]] - How to setup your PBX to participate in a DUNDi cloud.
* [[Conference calls]] - MeetMe simple conference rooms for Asterisk.
* [[Dimensioning Asterisk PBX]] - Dimensioning PBX-server.
* [[H323 configuration]] - How to setup your PBX to handle H.323 clients

== VoIP Services ==
* [[VoIPBuster]] - A provider that can connect VoIP-calls to the PSTN.
* [[PSGw]] - Personal Skype to SIP and H.323 gateway.

=== VoIP-test sites===
To test the quality or your VoIP connection you can use some web tools that give you some information

* [http://www.testyourvoip.com/ To test your VoIP quality and the MOS] (Mean Opinion Score, from 1 to 5) for some places around the world
* [http://www.bandwidth.com/tools/voipTest To test if you have any problem in your routers/firewalls] that do not let pass the VoIP communications.
* [http://www.talkswitch.com/voip/voip_capacity.php To know how many VoIP calls we can do simultaneously].

For comparison, see the [[VoIP-quality site testresults]].

== VoIP Protocols ==
* [[IAX2]] - Inter-Asterisk eXchange protocol
* [[DUNDi]] - Distributed Universal Number Discovery
* [[H323]]
* [[SIP vs IAX]]
* [[SIP vs H323]]

== Conference calls ==
* [http://www.sineapps.com/news.php?rssid=1667 Videoswitch support for MeetMe] - Very basic support for redirecting video within a MeetMe conference.
* [[app_conference]] - app_conference is a channel-independent conference application.

<big>[[Links#Conference_call_links|Conference call related links]]</big>.

== Analysis ==
<big>''Networking''</big>
* [[Wireshark]] - Successor of Ethereal
* [[Ethereal]] - Network traffic analyser
** [[Ethereal/Wireshark Analysis]] of VoIP traffic
** [[DUNDi network traffic]] - traffic analysis of DUNDi traffic.
*** [[Asterisk DUNDi debug]] - Analysis of DUNDi traffic with Asterisk's debug.
* [[Bandwidth estimates]]
** [[SIPp]] - test tool / traffic generator for the SIP protocol
* [[Controlling Bandwidth]]
<big>''Compression''</big>
* [[Audio Codecs]] - Coding/Decoding of audio streams
* [[Video Codecs]] - Coding/Decoding of video streams

== Conclusions ==

== [[Assignment 1: DUNDi & IAX (by Gijs & Ruben)]] ==

== [[Assignment 2: H323 (by Fabio & Anneke)]] ==
=== [[Sietse]] ===
=== [[Debug outputs]] ===

== Misc ==
* [[Documents]] - Collected documents
* [[Links]] - Various links with related info.
* [[DIY]] - Information on how to setup a cheap BPX yourself.
* [[Glossary]]
* [[Getting started]] with this Wiki.

== THALES ==
* [[THALES comments]] - THALES comments

VoIP Analysis - Revision history

Admin: Created page with "'''Analysis of VoIP''' == VoIP software == * Asterisk - A free PBX-system * Trixbox - A Linux-distro specially configured for Asterisk ** Trixbox 1.2.3 - O..."