Latest revision as of 08:48, 2 February 2017

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.

Desktop Integration

Click to call

click-to-call is a pharse used for functionality when a desktop user initiates VoIP-call from desktop software. There are different use cases when this could happen:

desktop application, that is linked to CLI binary with arguments: kaddressbook
web browsers clicking href=tel: that either have a plugin or CLI-helper that is executed with arguments. Related documentation: rfc3966, apple.com - phone links

Second part to click-to-call is targeted phone system, wether that is nearby PBX (Asterisk), a local desktop softphone or hardphone (Cisco SOAP/XML, etc).

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.
kphone interesting SIP-phone project for KDE that never had enough activity.
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
pycall is a flexible python library for creating and using Asterisk call files.
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 (This project is now called ring. https://ring.cx/ It depends (among other things) on openDHT for which a review request was open some time ago: 1377762)
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.

Applications/Libraries already packaged

asterisk - Open Source PBX
asterisk-sounds-core - core sounds for Asterisk
callweaver (former OpenPBX) - GPL-only fork of Asterisk.
ccrtp - Common C++ class framework for RTP/RTCP
ekiga - A Gnome based SIP/H323 teleconferencing application
hylafax+ - An enterprise-strength fax server
iax - Implementation of Inter-Asterisk eXchange protocol
iaxclient- Library for creating telephony solutions that interoperate with Asterisk
isdn4k-utils - Utilities for configuring an ISDN subsystem.
jabbin- Jabber and VoIP client (fork of well-known Psi)
jrtplib - C++ RTP library
kannel - WAP and SMS gateway
libeXosip2 - A library that hides the complexity of using the SIP protocol.
libosip2 - oSIP is an implementation of SIP.
libpri - An implementation of Primary Rate ISDN
libss7 - SS7 protocol services to applications
libzrtpcpp - ZRTP support library for the GNU ccRTP stack
linphone - Linphone is an internet phone or Voice Over IP phone (VoIP).
mISDN - Userspace part of Modular ISDN stack
nagios-plugins-check_sip - A Nagios plugin to check SIP servers and devices
opal - Open Phone Abstraction Library
openser - Fork of well-known SER SIP Server with interesting new features
opensips - Open Source SIP Server
openxcap - open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms.
ortp - A C library implementing the RTP protocol (RFC3550)
python-sippy - B2BUA SIP call controlling component
resiprocate - reSIProcate SIP stack, repro SIP proxy (with WebRTC support), reTurn ICE/STUN/TURN server, sipdialer (click to call)
rtpproxy - RTP proxy server
sems - an extensible SIP media server
ser - SIP Express Router
sip-redirect - Tiny IPv4 and IPv6 SIP redirect server written in Perl
sipp - test tool and traffic generator for the SIP protocol
sipsak - SIP swiss army knife
sipwitch - SIP telephony server for secure phone systems
sofia-sip - Sofia SIP UA Library
spandsp - A DSP library for telephony
stun - implements the stun protocol
xisdnload - An ISDN connection load average display for the X Window System
zaptel - Tools and libraries for using/configuring/monitoring Zapata telephony interfaces

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

Orphaned Packages Needing Maintainers

None, currently

Retired Packages

twinkle - SIP softphone - upstream dead, bitrot - [1] - still present in EPEL5 and EPEL6 as of 2014-05-29

Rejected Packages

MySTUN - No response from reporter
zaptel-kmod - kernel modules not allowed

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

References

@@ Line 6: / Line 6: @@
 == 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 [http://www.opentelecoms.org/federated-voip-quick-start-howto the Federated VoIP quick start HOWTO] for details.
+== Desktop Integration ==
+=== Click to call ===
+''click-to-call'' is a pharse used for functionality when a desktop user initiates VoIP-call from desktop software. There are different use cases when this could happen:
+* desktop application, that is linked to CLI binary with arguments: kaddressbook
+* web browsers clicking href=tel: that either have a plugin or CLI-helper that is executed with arguments. Related documentation: [https://tools.ietf.org/html/rfc3966 rfc3966], [https://developer.apple.com/library/ios/featuredarticles/iPhoneURLScheme_Reference/PhoneLinks/PhoneLinks.html apple.com - phone links]
+Second part to click-to-call is targeted phone system, wether that is nearby PBX (Asterisk), a local desktop softphone or hardphone (Cisco SOAP/XML, etc).
 == Applications/Libraries of Interest ==
@@ Line 16: / Line 28: @@
 * [http://www.gnugk.org/ 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.
+* [http://sourceforge.net/projects/kphone/ kphone] interesting SIP-phone project for KDE that never had enough activity.
 * [http://msrprelay.org/ MSRPRelay] - it helps in NAT traversal of media sessions between endpoints located behind NAT.
 * [http://chatserver.ag-projects.com/ SIP chatserver] - an open source conference bridge that supports MSRP chat sessions.
@@ Line 21: / Line 34: @@
 * [http://www.opensourcesip.org:8080/clearspacex/community/opensbc OpenSBC] - hybrid SIP proxy and B2BUA
 * [http://www.opensipstack.org/ OpenSIPStack] - implementation of the Session Initiation Protocol
+* [http://www.pycall.org/ pycall] '' is a flexible python library for creating and using Asterisk call files.''
 * [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://sflphone.org/ SFLphone] - the open-source enterprise-class SIP/IAX2 softphone
+* [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 (This project is now called ring. https://ring.cx/ It depends (among other things) on openDHT for which a review request was open some time ago: [[rhbug:1377762|1377762]])
 * [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://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.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.
+* [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.
+* [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 ==
@@ Line 37: / Line 53: @@
 * {{package|ccrtp}} - Common C++ class framework for RTP/RTCP
 * {{package|ekiga}} - A Gnome based SIP/H323 teleconferencing application
+* {{package|hylafax+}} - An enterprise-strength fax server
 * {{package|iax}} - Implementation of Inter-Asterisk eXchange protocol
 * {{package|iaxclient}}- Library for creating telephony solutions that interoperate with Asterisk
@@ Line 57: / Line 74: @@
 * {{package|ortp}} - A C library implementing the RTP protocol (RFC3550)
 * {{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|sems}} - an extensible SIP media server
@@ Line 67: / Line 85: @@
 * {{package|spandsp}} - A DSP library for telephony
 * {{package|stun}} - implements the stun protocol
-* {{package|twinkle}} - SIP softphone
 * {{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
 == 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
 ** 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/process_bug.cgi 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
+* [https://bugzilla.redhat.com/show_bug.cgi?id=1030307 Yate] - Yet Another Telephony Engine
 == Orphaned Packages Needing Maintainers ==
 * None, currently
+== Retired Packages ==
+* {{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 ==
@@ Line 97: / Line 117: @@
 * [[MatejCepl]]
 * [[Marionline]]
+* [[User:Pocock|Daniel Pocock]]
+* [[Tuju]]
 == General Issues ==
@@ Line 113: / Line 135: @@
 == References ==
 * [[Packaging/Guidelines#Legal]]
+* [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.voip-info.org/wiki/view/Open+Source+VOIP+Software Open Source VOIP applications]
 [[Category:Packaging SIGs]]

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