VoIP

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== Process ==
 
== Process ==
 
Want to suggest a VoIP application or library?  Just add it to the ''Applications/Libraries of Interest'' list.  Working on packaging a VoIP application or library, or need a review for your VoIP-related package?  Just add it to the ''Applications/Libraries Being Packaged'' list.  Interested in joining the SIG?  Just add your WikiName to the list.
 
Want to suggest a VoIP application or library?  Just add it to the ''Applications/Libraries of Interest'' list.  Working on packaging a VoIP application or library, or need a review for your VoIP-related package?  Just add it to the ''Applications/Libraries Being Packaged'' list.  Interested in joining the SIG?  Just add your WikiName to the list.
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== Federating VoIP and real-time communications with open standards ==
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You can help build a world of free communications based on secure, open standards by deploying SIP and XMPP on your Fedora and Red Hat servers.  See [http://www.opentelecoms.org/federated-voip-quick-start-howto the Federated VoIP quick start HOWTO] for details.
  
 
== Applications/Libraries of Interest ==
 
== Applications/Libraries of Interest ==
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** Initial attempt to package it was made, but review request was closed due to lack of activity
 
** Initial attempt to package it was made, but review request was closed due to lack of activity
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=226210 opal] - Open Phone Abstraction Library (merge review).
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=226210 opal] - Open Phone Abstraction Library (merge review).
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* [https://bugzilla.redhat.com/show_bug.cgi?id=892625 reSIProcate] - reSIProcate SIP stack, repro SIP proxy, reTurn ICE/STUN/TURN server, sipdialer (click to call)
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=509619 srtp] - Secure Real-Time Transport Protocol (SRTP) Library
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=509619 srtp] - Secure Real-Time Transport Protocol (SRTP) Library
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=728302 pjproject] - Libraries written in C language for building embedded/non-embedded VoIP applications
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=728302 pjproject] - Libraries written in C language for building embedded/non-embedded VoIP applications
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* [[MatejCepl]]
 
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== General Issues ==
 
== General Issues ==

Revision as of 18:53, 7 January 2013

Contents

VoIP Special Interest Group

Mission

To package as many Voice over IP applications as possible for Fedora. To that end, members of this SIG will assist in packaging VoIP applications and make reviewing VoIP-related packages our priority.

Process

Want to suggest a VoIP application or library? Just add it to the Applications/Libraries of Interest list. Working on packaging a VoIP application or library, or need a review for your VoIP-related package? Just add it to the Applications/Libraries Being Packaged list. Interested in joining the SIG? Just add your WikiName to the list.

Federating VoIP and real-time communications with open standards

You can help build a world of free communications based on secure, open standards by deploying SIP and XMPP on your Fedora and Red Hat servers. See the Federated VoIP quick start HOWTO for details.

Applications/Libraries of Interest

  • Asterisk-Addons
  • Asterisk-Sounds
  • Call Control - a prepaid application that can be used together with OpenSIPS call_control module and CDRTool rating engine to limit the duration of SIP sessions based on a prepaid balance. It can also be used to limit the duration of any session to a predefined maximum value without debiting a balance.
  • CDRTool - A set of utilities for working with call detail records
  • Druid - an open source web-based unified communications platform (based on Asterisk)
  • FreeSWITCH - an open source telephony platform.
  • gnugk - H.323 gatekeeper
    • This may require importing OpenH323 from the old Fedora package into Fedora Package Collection because Fedora dropped OpenH323 once GnomeMeeting/Ekiga switched to Opal.
  • MSRPRelay - it helps in NAT traversal of media sessions between endpoints located behind NAT.
  • SIP chatserver - an open source conference bridge that supports MSRP chat sessions.
  • minisip - SIP softphone
  • OpenSBC - hybrid SIP proxy and B2BUA
  • OpenSIPStack - implementation of the Session Initiation Protocol
  • QjSimple - cross-platform SIP Client, is based on the pjsip SIP stack and the Qt GUI toolkit.
  • QuteCom (former WengoPhone) - SIP compliant VoIP client
  • SFLphone - the open-source enterprise-class SIP/IAX2 softphone
  • SIP SIMPLE client - a Python software library that allows for easy development of Internet communications end-points based on SIP and related protocols for voice, rich presence, session based instant messaging (IM), file transfers and desktop sharing.
  • sipXecs - a SIP Unified Communications solution for your enterprise.
  • Yate - Yet Another Telephony Engine
  • Yxa - transaction stateful SIP stack and a set of SIP server applications
  • GreenJ - GreenJ is an open source Voice-over-IP phone software using pjsip and Qt
  • Homer conferencing - Homer is a free cross-platform SIP softphone, which also supports video conferencing. It can't be include in Fedora repository because it use ffmpeg. A review request is open on RPMFusion repository.
  • Jitsi - an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Applications/Libraries already packaged

  • Package-x-generic-16.pngasterisk - Open Source PBX
  • Package-x-generic-16.pngasterisk-sounds-core - core sounds for Asterisk
  • Package-x-generic-16.pngcallweaver (former OpenPBX) - GPL-only fork of Asterisk.
  • Package-x-generic-16.pngccrtp - Common C++ class framework for RTP/RTCP
  • Package-x-generic-16.pngekiga - A Gnome based SIP/H323 teleconferencing application
  • Package-x-generic-16.pngiax - Implementation of Inter-Asterisk eXchange protocol
  • Package-x-generic-16.pngiaxclient- Library for creating telephony solutions that interoperate with Asterisk
  • Package-x-generic-16.pngisdn4k-utils - Utilities for configuring an ISDN subsystem.
  • Package-x-generic-16.pngjabbin- Jabber and VoIP client (fork of well-known Psi)
  • Package-x-generic-16.pngjrtplib - C++ RTP library
  • Package-x-generic-16.pngkannel - WAP and SMS gateway
  • Package-x-generic-16.pnglibeXosip2 - A library that hides the complexity of using the SIP protocol.
  • Package-x-generic-16.pnglibosip2 - oSIP is an implementation of SIP.
  • Package-x-generic-16.pnglibpri - An implementation of Primary Rate ISDN
  • Package-x-generic-16.pnglibss7 - SS7 protocol services to applications
  • Package-x-generic-16.pnglibzrtpcpp - ZRTP support library for the GNU ccRTP stack
  • Package-x-generic-16.pnglinphone - Linphone is an internet phone or Voice Over IP phone (VoIP).
  • Package-x-generic-16.pngmISDN - Userspace part of Modular ISDN stack
  • Package-x-generic-16.pngnagios-plugins-check_sip - A Nagios plugin to check SIP servers and devices
  • Package-x-generic-16.pngopal - Open Phone Abstraction Library
  • Package-x-generic-16.pngopenser - Fork of well-known Package-x-generic-16.pngSER SIP Server with interesting new features
  • Package-x-generic-16.pngopensips - Open Source SIP Server
  • Package-x-generic-16.pngopenxcap - open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms.
  • Package-x-generic-16.pngortp - A C library implementing the RTP protocol (RFC3550)
  • Package-x-generic-16.pngpython-sippy - B2BUA SIP call controlling component
  • Package-x-generic-16.pngrtpproxy - RTP proxy server
  • Package-x-generic-16.pngsems - an extensible SIP media server
  • Package-x-generic-16.pngser - SIP Express Router
  • Package-x-generic-16.pngsip-redirect - Tiny IPv4 and IPv6 SIP redirect server written in Perl
  • Package-x-generic-16.pngsipp - test tool and traffic generator for the SIP protocol
  • Package-x-generic-16.pngsipsak - SIP swiss army knife
  • Package-x-generic-16.pngsipwitch - SIP telephony server for secure phone systems
  • Package-x-generic-16.pngsofia-sip - Sofia SIP UA Library
  • Package-x-generic-16.pngspandsp - A DSP library for telephony
  • Package-x-generic-16.pngstun - implements the stun protocol
  • Package-x-generic-16.pngtwinkle - SIP softphone
  • Package-x-generic-16.pngxisdnload - An ISDN connection load average display for the X Window System
  • Package-x-generic-16.pngzaptel - Tools and libraries for using/configuring/monitoring Zapata telephony interfaces

Packages for review

  • HylaFAX - is a enterprise-strength fax server supporting Class 1 and 2 fax modems on UNIX systems.
  • Mediaproxy - far-end NAT traversal solution for SER/OpenSER
    • Initial attempt to package it was made, but review request was closed due to lack of activity
  • opal - Open Phone Abstraction Library (merge review).
  • reSIProcate - reSIProcate SIP stack, repro SIP proxy, reTurn ICE/STUN/TURN server, sipdialer (click to call)
  • srtp - Secure Real-Time Transport Protocol (SRTP) Library
  • pjproject - Libraries written in C language for building embedded/non-embedded VoIP applications

Orphaned Packages Needing Maintainers

  • None, currently

Rejected Packages

Packagers/Reviewers/People interested

General Issues

* Legal status of iLBC prevents it from inclusion into Fedora, so you should check that your application doesn't use it before submitting it to Fedora/EPEL.

Now iLBC can be included in Fedora: https://bugzilla.redhat.com/show_bug.cgi?id=728302#c26

A new review request for WebRTC, that include iLBC codec, should be open nearly. A first package and spec file can be found here: https://bugzilla.redhat.com/show_bug.cgi?id=728302#c41

See also

References