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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* [http://www.ipcom.at/index.php?id=560 QjSimple] - cross-platform SIP Client, is based on the pjsip SIP stack and the Qt GUI toolkit.
 
* [http://www.ipcom.at/index.php?id=560 QjSimple] - cross-platform SIP Client, is based on the pjsip SIP stack and the Qt GUI toolkit.
 
* [http://www.qutecom.org/ QuteCom] (former WengoPhone) - SIP compliant VoIP client
 
* [http://www.qutecom.org/ QuteCom] (former WengoPhone) - SIP compliant VoIP client
 +
* [http://www.resiprocate.org reSIProcate] - comprehensive implementation of a SIP stack in C++, repro SIP proxy (with WebRTC support), reTurn ICE/STUN/TURN server, sipdialer (click to call) - supports IPv6, TLS, WebSockets, Python-based routing scripts, low-level and high-level programming APIs
 
* [http://sflphone.org/ SFLphone] - the open-source enterprise-class SIP/IAX2 softphone
 
* [http://sflphone.org/ SFLphone] - the open-source enterprise-class SIP/IAX2 softphone
 
* [http://sipsimpleclient.com/ 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.
 
* [http://sipsimpleclient.com/ 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.
 
* [http://sipxecs.sipfoundry.org/ sipXecs] - a SIP Unified Communications solution for your enterprise.
 
* [http://sipxecs.sipfoundry.org/ sipXecs] - a SIP Unified Communications solution for your enterprise.
* [http://yate.null.ro/pmwiki/index.php?n=Main.Download Yate] - Yet Another Telephony Engine
 
 
* [http://www.stacken.kth.se/project/yxa/ Yxa] - transaction stateful SIP stack and a set of SIP server applications
 
* [http://www.stacken.kth.se/project/yxa/ Yxa] - transaction stateful SIP stack and a set of SIP server applications
 
* [http://www.loremipsum.at/produkte/greenj/ GreenJ] - GreenJ is an open source Voice-over-IP phone software using pjsip and Qt
 
* [http://www.loremipsum.at/produkte/greenj/ GreenJ] - GreenJ is an open source Voice-over-IP phone software using pjsip and Qt
 
* <del>[http://www.homer-conferencing.com Homer conferencing] - Homer is a free cross-platform SIP softphone, which also supports video conferencing.</del> It can't be include in Fedora repository because it use ffmpeg. A review request is open on RPMFusion repository.
 
* <del>[http://www.homer-conferencing.com Homer conferencing] - Homer is a free cross-platform SIP softphone, which also supports video conferencing.</del> It can't be include in Fedora repository because it use ffmpeg. A review request is open on RPMFusion repository.
 
* [https://jitsi.org/index.php/Main/HomePage 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.
 
* [https://jitsi.org/index.php/Main/HomePage 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.
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* [http://icanblink.com/ Blink] - Blink is the real-time communications client using SIP protocol. You can use it with many SIP providers, on the LAN using Bonjour and SIP2SIP free service.
  
 
== Applications/Libraries already packaged ==
 
== Applications/Libraries already packaged ==
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* {{package|ccrtp}} - Common C++ class framework for RTP/RTCP
 
* {{package|ccrtp}} - Common C++ class framework for RTP/RTCP
 
* {{package|ekiga}} - A Gnome based SIP/H323 teleconferencing application
 
* {{package|ekiga}} - A Gnome based SIP/H323 teleconferencing application
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* {{package|hylafax+}} - An enterprise-strength fax server
 
* {{package|iax}} - Implementation of Inter-Asterisk eXchange protocol
 
* {{package|iax}} - Implementation of Inter-Asterisk eXchange protocol
 
* {{package|iaxclient}}- Library for creating telephony solutions that interoperate with Asterisk
 
* {{package|iaxclient}}- Library for creating telephony solutions that interoperate with Asterisk
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* {{package|ortp}} - A C library implementing the RTP protocol (RFC3550)
 
* {{package|ortp}} - A C library implementing the RTP protocol (RFC3550)
 
* {{package|python-sippy}} - B2BUA SIP call controlling component
 
* {{package|python-sippy}} - B2BUA SIP call controlling component
 +
* {{package|resiprocate}} - reSIProcate SIP stack, repro SIP proxy (with WebRTC support), reTurn ICE/STUN/TURN server, sipdialer (click to call)
 
* {{package|rtpproxy}} - RTP proxy server
 
* {{package|rtpproxy}} - RTP proxy server
 
* {{package|sems}} - an extensible SIP media server
 
* {{package|sems}} - an extensible SIP media server
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* {{package|spandsp}} - A DSP library for telephony
 
* {{package|spandsp}} - A DSP library for telephony
 
* {{package|stun}} - implements the stun protocol
 
* {{package|stun}} - implements the stun protocol
* {{package|twinkle}} - SIP softphone
 
 
* {{package|xisdnload}} - An ISDN connection load average display for the X Window System
 
* {{package|xisdnload}} - An ISDN connection load average display for the X Window System
 
* {{package|zaptel}} - Tools and libraries for using/configuring/monitoring Zapata telephony interfaces
 
* {{package|zaptel}} - Tools and libraries for using/configuring/monitoring Zapata telephony interfaces
  
 
== Packages for review ==
 
== Packages for review ==
* [https://bugzilla.redhat.com/show_bug.cgi?id=188542 HylaFAX] - is a enterprise-strength fax server supporting Class 1 and 2 fax modems on UNIX systems.
 
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=525412 Mediaproxy] - far-end NAT traversal solution for SER/OpenSER
 
* [https://bugzilla.redhat.com/show_bug.cgi?id=525412 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
 
** 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=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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* [https://bugzilla.redhat.com/show_bug.cgi?id=1030307 Yate] - Yet Another Telephony Engine
  
 
== Orphaned Packages Needing Maintainers ==
 
== Orphaned Packages Needing Maintainers ==
  
 
* None, currently
 
* None, currently
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== Retired Packages ==
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* {{package|twinkle}} - SIP softphone - upstream dead, bitrot - [http://lists.fedoraproject.org/pipermail/devel/2013-March/179593.html] - still present in EPEL5 and EPEL6 as of 2014-05-29
  
 
== Rejected Packages ==
 
== Rejected Packages ==
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* [[MatejCepl]]
 
* [[MatejCepl]]
 
* [[Marionline]]
 
* [[Marionline]]
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* [[User:Pocock|Daniel Pocock]]
  
 
== General Issues ==
 
== General Issues ==
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== References ==
 
== References ==
 
* [[Packaging/Guidelines#Legal]]
 
* [[Packaging/Guidelines#Legal]]
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* [http://rtcquickstart.org Real-Time Communications (RTC) quick start guide]
 
* [http://www.redhatmagazine.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ Open source telephony: a Fedora-based VoIP server with Asterisk]
 
* [http://www.redhatmagazine.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ Open source telephony: a Fedora-based VoIP server with Asterisk]
 
* [http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software Open Source VOIP applications]
 
* [http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software Open Source VOIP applications]
  
 
[[Category:Packaging SIGs]]
 
[[Category:Packaging SIGs]]

Latest revision as of 09:26, 17 July 2014

Contents

[edit] VoIP Special Interest Group

[edit] 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.

[edit] 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.

[edit] 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.

[edit] 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
  • reSIProcate - comprehensive implementation of a SIP stack in C++, repro SIP proxy (with WebRTC support), reTurn ICE/STUN/TURN server, sipdialer (click to call) - supports IPv6, TLS, WebSockets, Python-based routing scripts, low-level and high-level programming APIs
  • 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.
  • 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.
  • Blink - Blink is the real-time communications client using SIP protocol. You can use it with many SIP providers, on the LAN using Bonjour and SIP2SIP free service.

[edit] 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.pnghylafax+ - An enterprise-strength fax server
  • 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.pngresiprocate - reSIProcate SIP stack, repro SIP proxy (with WebRTC support), reTurn ICE/STUN/TURN server, sipdialer (click to call)
  • 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.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

[edit] Packages for review

  • 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
  • srtp - Secure Real-Time Transport Protocol (SRTP) Library
  • pjproject - Libraries written in C language for building embedded/non-embedded VoIP applications
  • Yate - Yet Another Telephony Engine

[edit] Orphaned Packages Needing Maintainers

  • None, currently

[edit] Retired Packages

  • Package-x-generic-16.pngtwinkle - SIP softphone - upstream dead, bitrot - [1] - still present in EPEL5 and EPEL6 as of 2014-05-29

[edit] Rejected Packages

[edit] Packagers/Reviewers/People interested

[edit] 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

[edit] See also

[edit] References