Latest revision as of 18:53, 7 January 2013

[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
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.

[edit] 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
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
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
twinkle - SIP softphone
xisdnload - An ISDN connection load average display for the X Window System
zaptel - Tools and libraries for using/configuring/monitoring Zapata telephony interfaces

[edit] 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

[edit] Orphaned Packages Needing Maintainers

None, currently

[edit] Rejected Packages

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

[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

Telepathy

[edit] References

@@ Line 2: / Line 2: @@
 == Mission ==
-To package as many Voice over IP applications as possible for Fedora.<BR>
+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.
-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 [http://www.opentelecoms.org/federated-voip-quick-start-howto the Federated VoIP quick start HOWTO] for details.
 == Applications/Libraries of Interest ==
-* [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://www.minisip.org/ minisip] - SIP softphone
-* [http://www.xs4all.nl/~mfnboer/twinkle/index.html twinkle] - SIP softphone
 * Asterisk-Addons
 * Asterisk-Sounds
-* [http://www.iptel.org/sems SEMS] - SIP express media server
+* [http://callcontrol.ag-projects.com/ 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.
-* [http://mediaproxy.ag-projects.com/ Mediaproxy] - far-end NAT traversal solution for SER/OpenSER
 * [http://cdrtool.ag-projects.com/ CDRTool] - A set of utilities for working with call detail records
-* [http://openwengo.com/ OpenWengo] - SIP softphone with lot of advanced features
+* [http://www.voiceroute.org/ Druid] - an open source web-based unified communications platform (based on Asterisk)
-** This softphone uses a lot of other libraries (as optional and as a required) and there are not allowed for inclusion in Fedora among them.
+* [http://www.freeswitch.org/ FreeSWITCH] - an open source telephony platform.
+* [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://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.
+* [http://www.minisip.org/ minisip] - SIP softphone
+* [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.opensourcesip.org/opensbc.php OpenSBC] - hybrid SIP proxy and B2BUA
+* [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://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.jabbin.com/int/ Jabbin] - Jabber and VoIP client (fork of well-known Psi)
+* [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.
 == Applications/Libraries already packaged ==
-* [http://openser.org OpenSER] - Fork of well-known SER SIP Server with interesting new features
+* {{package|asterisk}} - Open Source PBX
-* [http://rtpproxy.org/wiki/RTPproxy rtpproxy] - RTP proxy server
+* {{package|asterisk-sounds-core}} - core sounds for Asterisk
-* [http://sipp.sourceforge.net/ SIPp] - test tool and traffic generator for the SIP protocol
+* {{package|callweaver}} (former OpenPBX) - GPL-only fork of Asterisk.
-* [http://www.callweaver.org/ CallWeaver] (former OpenPBX) - GPL-only fork of Asterisk.
+* {{package|ccrtp}} - Common C++ class framework for RTP/RTCP
-* Asterisk - Open Source PBX
+* {{package|ekiga}} - A Gnome based SIP/H323 teleconferencing application
-* SER - SIP Express Router
+* {{package|iax}} - Implementation of Inter-Asterisk eXchange protocol
-* sipsak - SIP swiss army knife
+* {{package|iaxclient}}- Library for creating telephony solutions that interoperate with Asterisk
-* SIPp - SIP test tool / traffic generator
+* {{package|isdn4k-utils}} - Utilities for configuring an ISDN subsystem.
-* sofia-sip - Sofia SIP UA Library
+* {{package|jabbin}}- Jabber and VoIP client (fork of well-known Psi)
-* ekiga
+* {{package|jrtplib}} - C++ RTP library
-* spandsp
+* {{package|kannel}} - WAP and SMS gateway
-* zaptel
+* {{package|libeXosip2}} - A library that hides the complexity of using the SIP protocol.
-* libpri
+* {{package|libosip2}} - oSIP is an implementation of SIP.
-* jrtplib - C++ RTP library
+* {{package|libpri}} - An implementation of Primary Rate ISDN
+* {{package|libss7}} - SS7 protocol services to applications
+* {{package|libzrtpcpp}} - ZRTP support library for the GNU ccRTP stack
+* {{package|linphone}} - Linphone is an internet phone or Voice Over IP phone (VoIP).
+* {{package|mISDN}} - Userspace part of Modular ISDN stack
+* {{package|nagios-plugins-check_sip}} - A Nagios plugin to check SIP servers and devices
+* {{package|opal}} - Open Phone Abstraction Library
+* {{package|openser}} - Fork of well-known {{package|ser|SER}} SIP Server with interesting new features
+* {{package|opensips}} - Open Source SIP Server
+* {{package|openxcap}}  - open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms.
+* {{package|ortp}} - A C library implementing the RTP protocol (RFC3550)
+* {{package|python-sippy}} - B2BUA SIP call controlling component
+* {{package|rtpproxy}} - RTP proxy server
+* {{package|sems}} - an extensible SIP media server
+* {{package|ser}} - SIP Express Router
+* {{package|sip-redirect}} - Tiny IPv4 and IPv6 SIP redirect server written in Perl
+* {{package|sipp}} - test tool and traffic generator for the SIP protocol
+* {{package|sipsak}} - SIP swiss army knife
+* {{package|sipwitch}} - SIP telephony server for secure phone systems
+* {{package|sofia-sip}} - Sofia SIP UA Library
+* {{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/bugzilla/show_bug.cgi?id=189949 MySTUN] - STUN server
+* [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=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=728302 pjproject] - Libraries written in C language for building embedded/non-embedded VoIP applications
 == Orphaned Packages Needing Maintainers ==
+* None, currently
 == Rejected Packages ==
+* [https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=189949 MySTUN] - No response from reporter
 * [https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=177583 zaptel-kmod] - kernel modules not allowed
@@ Line 58: / Line 101: @@
 * [[PeterLemenkov]]
 * [[DavidWoodhouse]]
+* [[MatejCepl]]
+* [[Marionline]]
+* [[DanielPocock]]
 == General Issues ==
+<del>* 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.</del>
+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 ==
+* [[Telepathy]]
 == References ==
-* http://fedoraproject.org/wiki/Packaging/Guidelines#Legal
+* [[Packaging/Guidelines#Legal]]
+* [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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