开源VOIP软件

Open Source VOIP Software

Open Source VOIP applications, both clients and servers.

Open source means all source code is available!! Do not post any "free but not open" software here!

SIP Proxies

Sip I/O Lightweight sip proxy, location server, and registrar
SBO SIP Proxy Bypass
All types of Internet Firewall
JAIN-SIP
Proxy
Mini-SIP-Proxy A
very tiny perl POE based SIP proxy
MjServer cross-platform
SIP proxy/registrar/redirect, written in java, based on MjSip stack
MySIPSwitch SIP
Proxy server which allows using multiple SIP accounts with a single SIP login
NethidPro3.0.6 Opensource
Sip Encryption Bridge: www.vonets.com
Net-SIP A
Perl SIP framework that includes a stateless proxy
OpenJSIP Opensource
distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. Based on NIST
SIP and derived from JAIN-SIP Proxy.
OpenSBC:
MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM
OpenSER:
GPL SIP Server with TLS support - renamed toKamailio
OpenSIPS forked
from OpenSER.
partysip SIP
proxy server
repro
from the reSIProcate project fully implementsFederated VoIP and
has a built-in web UI for quick setup
REMWAVE Calamar Cross-platform
high performance SIP proxy written in Java
SaRP SIP
and RTP Proxy in Perl
SIP
Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org
Siproxd SIP
and RTP Proxy
SIPVicious tool
suite: tools for auditing sip devices
sipX The
SIP PBX for Linux: Complete, native SIP PBX solution for business
Vocal SIP
softswitch with H.323 and MGCP translators for non-SIP endpoints
Yxa Written
in the Erlang programming language
CRM INtegration Proxy Open Source program writen on java. based on MJ SIP lib Proxy for Call-Centers solutions
Clearwater -
open source IMS (IP Multimedia Subsystem) implementation designed for massively scalable deployment in the Cloud - SIP routing components built on PJSIP

SIP Clients (UA's)

Android clients:

Brief
Msg is simple SIP messenger.
Lumicall is
a heavily enhanced derivative of SIPdroid, adding support for ZRTP, SRTP, ENUM, ICE/TURN
SIPdroid is
a basic SIP dialer for Android, based on the MjSIP stack in Java
CSIPSimple is
an alternative SIP dialer for Android, based on the PJSIP C-implementation of SIP/RTP
ENUMdroid is
an ENUM lookup tool for Android's dialer, it relies on the user having some other softphone installed to make the call over SIP or Jabber
Sipmobile is
an opensource VoIP client for Android. Supports OPUS and VP8 codecs, Google push notifications, picture sharing. Setting are optimized for use with sipmobile.org domain. Can be used with another proxies.

Linux clients:

SBO
Multipath with Integrated SyncSwitch- Linux based SIP Solution.
Baresip Portable
SIP useragent with Video support
Blink: It
supports wideband VoIP (Opus codec), Chat, File Transfer and Multiparty conferencing based on MSRP protocol
Cockatoo
Ekiga || SIP, H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
FreeSWITCH:
Console client for SIP, IAX2, Woomera and Jingle/Google Talk
Jitsi (formerly
SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
Kphone
Homer
- live conferencing and more: free cross-platform video conferencing software, supporting unlimited amount of participants in a video/audio conference
Linphone audio
and video SIP softphone for Linux and Windows XP
minisip cross-platform
SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
MUVConf cross-platform
SIP multi-user video conference. See demo video. Download from code.google.com
MjUA: simple
cross-platform SIP softphone, written in java, based on MjSip stack
Open IP
Phone Business IP Phone sdk support, ims compliant, good interoperability.
OpenSIPStack MPL
licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
OpenSoftphone:
A simple Java based SIP softphone using the PjSip-jni wrapper.
OpenZoep:
GPL telephone and IM messaging client engine
Peers Minimalist
SIP softphone written in java (tested on linux and windows)
PhoneGaim
PJSUA:
Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo:
a fully SIP compliant multiplatform softphone with many features
SFLphone,
open-source multiplatform multi-protocol VoIP client
Shtoom: SIP softphone
in Python, runs on Windows, Mac, Linux
SipToSis from mhspot.com Skype
SIP UA - Multiplatform - Open Source
sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
sipXphone from SIPfoundry, previously known as the Pingtel phone
Twinkle
YateClient is multiprotocol
and multiplatform softphone with H.323, SIP, Jingle and IAX support.
YeaPhone:
A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
CRM Integration Client Open Source program writen on java. based on MJ SIP and SIP-Communicator for Call-Centers solutions

MacOS X clients:

Blink: It
supports wideband VoIP, Instant Messaging, File Transfer and Desktop Sharing based on MSRP
FreeSWITCH:
Console client for SIP, IAX2, Woomera and Jingle/Google Talk
Jitsi (formerly
SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
PJSUA:
Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo:
a fully SIP compliant multiplatform softphone with many features
SFLphone,
open-source multiplatform multi-protocol VoIP client
Shtoom: SIP softphone
in Python, runs on Windows, Mac, Linux
SipToSis from http://www.mhspot.com Skype
SIP UA - Multiplatform - Open Source
Telephone:
A SIP softphone designed for the Mac (written in Objective-C/Cocoa). Very good integration with Mac OSX : Dial from Addressbook, dial tel: URIs from Safari, notifications with Growl.
YateClient skinnable
VoIP client based on QT library which supports H.323, SIP, Jingle and IAX protocols
REMWAVE Communicator
OS X Open source SIP phone for OS X. Based on PJSIP library, scriptable with Apple Script and address book integration.

Windows clients

Blink: It
supports wideband VoIP (Opus codec), Chat, File Transfer and Multiparty conferencing based on MSRP protocol
Brief Msg is
simple SIP messenger.
Ekiga || SIP, H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. FormerlyGnomeMeeting
FreeSWITCH:
Console client for SIP, IAX2, Woomera and Jingle/Google Talk
Homer
- live conferencing and more: free cross-platform video conferencing software, supporting unlimited amount of participants in a video/audio conference
Jitsi (formerly
SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
JPhone Rich
software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
Linphone audio
and video SIP softphone for Linux and Windows XP
MicroSIP:
lightweight SIP softphone based on PJSIP stack for Windows OS written in C++. SIMPLE IM and Presense.
minisip cross-platform
SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
MUVConf cross-platform
SIP multi-user video conference. See demo video. Download from code.google.com
MjUA: simple
cross-platform SIP softphone, written in java, based on MjSip stack
OfficeSIP
Messenger is audio-video softphone and instant messenger, open
source alternative to MS Office Communicator.
OfficeSIP
Softphone GPL audio-video softphone.
OpenSIPStack MPL
licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
OpenSoftphone:
A simple Java based SIP softphone using the PjSip-jni wrapper
OpenZoep:
GPL telephone and IM messaging client engine
Peers Minimalist
SIP softphone written in java (tested on linux and windows)
PhoneGaim
PJSUA:
Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo:
a fully SIP compliant multiplatform softphone with many features
REMWAVE Communicator
Win Open source soft phone for Windows. Written in C# and based on the PJSIP library. Including branding engine.
Shtoom: SIP softphone
in Python, runs on Windows, Mac, Linux
SipToSis from mhspot.com Skype
SIP UA - Multiplatform - Open Source
sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
sipXphone from SIPfoundry, previously known as the Pingtel phone
tSIP Portable,
BSD-licensed softphone with BLF, call recording, customizable keypad and shortcuts, browser integration. Based on re/rem/baresip.
VMukti
(formerly 1videoConference) alpha: a web2.0 VoIP video conferencing software for Asterisk.
wxCommunicator Windows
softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support
YateClient is multiprotocol
and multiplatform softphone with H.323, SIP,Jingle and IAX support.

Platform independent clients

GreenJ:
Development framework based on Qt and PJSIP for easily building SIP-Softphone applications with a Web-Interface.
MUVConf cross-platform
SIP multi-user video conference. See demo video. Download from code.google.com
Weavver
Browser Phone: A web-browser based soft phone that's easy to integrate with any website. Works with the RTMP protocol as integrated in FreeSwitch. You can use this Flash-based front end with FreeSwitch to reach nearly any VoIP back-end (SIP/H.323/IAX/etc).

SIP tools

Callflow:
Generates SIP Call Flow diagrams
miTester
for SIP: SIP testing tool; Automates test execution.
Open Source Asterisk AMI: Open Source Asterisk AMI interface application
pjsip-perf:
SIP transaction and call performance measurement tool
PROTOS
Test-Suite: SIP Testing tools
SFTF: SIP Forum Test Framework - a SIP UA test suite primarily targeted at UA software developers hosted by SIPfoundry
SIP-CallerID:
SIP Caller ID retrieval and lookup
SIPbomber:
SIP proxy testing tool
SIP
SIMPLE Command Line Tools for SIP sessions (complete console based SIP UA) and SIMPLE Presence (Publish, Subscribe, Notify) and XCAP document manipulation
SIPInspector -
SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap file. Transport
protocols: UDP, TCP, websocket
Sipp: SIP
performance tester
Sipper:
SIPr (called Sipper) is an open source and a comprehensive SIP application testing framework. Generate any call flow in minutes.
SIP
Proxy: SIP security testing tool.
Sipsak:
SIP testing tool
SIP Soft
client: Software development kit for SIP Softphone
SIPVicious
tool suite: tools for auditing SIP devices
Vovida.org
load balancer: SIP Load Balancer

SIP Protocol Stacks and Libraries

Aloha Spring based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models
eXosip -
eXtended osip library
Juphoon
SIP Stack Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
libdissipate SIP stack
Libre -
Portable SIP Stack under BSD license with IPv4/v6 support (SIP,SDP,RTP/RTCP,STUN,TURN,ICE,DNS)
minisip includes
a SIP stack
MjSip -
complete and powerful java-based SIP library for both J2SE and J2ME platforms.
MSRP
Library - MSRP protocol (RFC4975) and its relay extension (RFC4976) written in Python
NIST
SIP Various SIP appications and tools in Java
Open
Sip Stack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
oSIP
Library SIP Library
OSP
client protocol stack and SIPfoundry
PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity
PJSIP:
Small footprint, high performance, and ultra-portable SIP stack written in C, and has language binding for Python. Works on smartphones (Symbian, Windows, iPhone/iOS, Android) as well as desktops and support ZRTP encryption.
reSIProcate SIP
stack and sample Application from SIPfoundry
SailFin Adds
SIP support the the Java GlassFish Application Server
SIP.js -
SIP Signaling JavaScript Library for WebRTC Developers
sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry
http://sofia-sip.sourceforge.net Sofia-Sip
is SIP stack implementation with STUN and presense support
SIP SIMPLE
client SDK - High level middleware on top of SIP, RTP, MSRP and XCAP protocols
Twisted Python
protocol stacks and applications includes SIP support
Verona - GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X
Vovida
SIP Vovida SIP stack
XCAP
Library - XCAP client library written in Python
YASS - Statefull SIP stack used in Yate written
in C++ usable for client, server or proxy in a multithread or single thread model. It's working on both Windows and Linux, it's very small but full featured.
ivrworx -
high level Lua interface to SIP/RTSP/MRCP, for testing distributed VoIP scenarios (windows, Vista+ clients).

H.323 Clients

Linux clients:

Ekiga || SIP, H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. FormerlyGnomeMeeting
FreeSWITCH:
Console client using OPAL
GnomeMeeting
YateClient is multiprotocol
and multiplatform softphone with H.323, SIP and IAX support.

MacOS X clients:

FreeSWITCH:
Console client using OPAL
ohphoneX
YateClient skinnable
VoIP client based on the QT library which supports H.323, SIP, Jingle and IAX protocols

Windows clients:

Ekiga || SIP, H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. FormerlyGnomeMeeting
FreeSWITCH:
Console client using OPAL
OpenPhone
YateClient is multiprotocol
and multiplatform softphone with H.323, SIP and IAX support.

H.323 Gatekeeper

GNU Gatekeeper -
for Linux, Windows, Mac etc.

IAX clients

FreeSWITCH
IAXComm for
Linux, MacOS X and Windows
Kiax - for
Linux, Windows and MacOS, based on iaxclient, GPL
MozIAX
QtIax from http://www.holgerschurig.de/qtiax.html
SFLphone,
open-source multiplatform multi-protocol VoIP client (IAX support is planned)
YakaPhoneSimple,
Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware
YateClient is multiprotocol
and multiplatform softphone with H.323, SIP and IAX support.

TURN servers and RTP Proxies

reTurn from
the reSIProcate project provides a standards compliant STUN/TURN relay
STUN
& TURN Server - is an open source STUN & TURN Server (and client library) for UNIX/Linux platforms.
AG
Projects: MediaProxy 1 works with SIP express router and OpenSER,
has load-balancing using DNS SRV records and accounting capabilities
Maxim
Sobolev RTPproxy: Works with SIP express router to
traverse NAT, also functions as RTP gateway between IPv4 and IPv6
MediaProxy
2 is more scalable using kernel space switching and works with OpenSIPs

RTP Protocol Stacks

ccRTP C++
library based on GNU Common C++
Juphoon
RTP Stack Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
JRTPLIB C++
object oriented RTP library
libRTP part of gnome-o-phone
libzrtpcpp -
ZRTP extension library for ccRTP stack
LIVE.COM
Streaming Media includes C++ RTP stack
oRTP Written
in C, running on linux, win32 and arm-linux.
PJMEDIA:
Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment
RTPlib C library
sipXmediaLib RTP + audio bridges, audio splitters, echo suppression, tone from generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from SIPfoundry
Secure RTP - see: SRTP
OpenTelecoms.org
ZRTP stack implements ZRTP in Java, for Android, J2SE and Blackberry, used in the Lumicall dialer
for Android
UCL
Common Multimedia Library includes cross platform RTP stack
Vovida
RTP Stack
YRTP - Yate RTP
stack, that can be used in other projects.
zrtp4j -
ZRTP stack for Java, based on GNU ZRTP, used in Jitsi (formerly SIP Communicator)

MSRP Relays

MSRPRelay from AG
Projects

XCAP servers

OpenXCAP from AG
Projects

Other tools

Encours Teleconferencing
in your web browser with an integrated VOIP layer (Java) and an optional Asterisk connectivity on the server side.
Howler
Technologies - optimised G.729 codec for softswitch market.
Interactive
Dialplanner Open-Source GUI Dialplan Development for Asterisk PBX.
MORCC -
automated online Calling Card store. Paypal integrated.
OfficeSIP
Turn Server is open source TURN server compatible
with |MS-TURN| extension.
OgonPhonesXML .NET
Library for Aastra SIP Phones and Cisco SIP/IP phones for fast and easy XML Interfacement.
OpenBTS A
Unix VOIP interface to the GSM cellular network
Oreka capture
and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with audio compression, rdbms metadata storage and web based user interface.
outCALL integrates
Microsoft's Outlook with Asterisk for pop-ups and click2dial. For Exchange integration see outCALL.
TBDialOut is
a Thunderbird extension that adds clickable links, context menu options and toolbar buttons to Thunderbird's address book, enabling you to dial direct from your addressbook. TBDialOut can be used with most softphones, with Snom, Yealink and Tiptel hardware
phones, with some Cisco systems and with Asterisk.
Vovida.org
STUN server: A STUN server
Voipong -
Voice over IP (VoIP) sniffer and call detector.
Vomit converts
a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file

PBX platforms

Some of these include SIP proxy functionality
Asterisk:
Open Source PBX. Supports IAX, SIP, MGCP, H.323 and
other protocols
CallWeaver:
a fork of Asterisk with T.38 termination
Elastix Unified
Communications distro supporting IP-PBX and Soft Switch capabilities
FreeSWITCH Open
Source PBX and Soft Switch
OpenPBX:
Open Source PBX developed using Perl
ZULTYS:
Open Standards PBX based on SIP
PBX4Linux: ISDN PBX with H.323 GW
sipwitch:
GNU project's Pure SIP call server, sipwitch on freshmeat.net
sipX - The SIP PBX for Linux from SIPfoundry http://sipxcom.org/ \sipXcom - Open Source Enterprise-ready full PBX replacement
SIP -
It's the Rage! - Rage! Business Office Xchange based on SipFoundry
YATE Yet Another
Telephony Engine - supports H.323, SIP, IAX,
PSTN

IVR platforms

Asterisk:
Open Source PBX with built-in IVR server
Bayonne:
GNU project IVR server
CT
Server Perl based Open Source client/server library supporting Voicetronix Telephony hardware.
Elastix Unified
Communications distro supporting IVR capabilities
FreeSWITCH
ICTDialer An
Open Source smart autodialer software bundled with graphical IVR Designer tools.
OpenVXI:
Implementation of VoiceXML
SEMS: Free/Open
Source SIP media server with IVR capabilities
sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant) http://sipxcom.org/ \sipXcom - Open Source Enterprise-ready extensive IVR support
YATE Yet Another
Telephony Engine
See Also: VoiceXML

Voice broadcasting platform

Newfies-Dialer
Open Source Autodialer & Voice Broadcasting Solution - Multi-Tenant system comprising Auto-dialer, survey tool, extension dialing (press 1 campaign), voice recording and Do Not Call, with white labeling, SMS and AMD available.
ICTDialer Is
an Open Source unified communications autodialer and broadcasting software application supporting voice, sms, fax broadcasting.

Voicemail servers

Asterisk:
Open Source PBX with built-in Voicemail Server
Elastix Unified
Communications distro supporting Voicemail capabilities
FreeSWITCH
Lintad:
Linux Telephone Answering Device - A Voice and Faxmail Server
OpenPBX:
Open Source PBX with built in voicemail
OpenUMS:
Linux Voicemail and Unified Messaging Server
SEMS: Free/Open
Source SIP media server with built-in Voicemail and Voicebox Server
sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
VOCP:
A Voicemail Server for voice modems
YATE Yet Another
Telephony Engine with H.323, SIP and IAX support.

Speech

Text-to-speech and speech-to-text (voice recognition)
Festival:
Voice synthesis system (implemented with a trainable neural network)
OpenSALT:
Implementation of SALT
OpenVXI:
Implementation of VoiceXML
Sphinx:
speaker-independent speech recognizer
UniMRCP:
cross-platform MRCP client and server

SMS solutions

jSMPP: low-level
Java API for SMPP, the protocol for SMS gateways on the Internet
SMS Router:
server process for handling interchange of SMS messages between an SMPP gateway and local applications using JMS, STOMP, SIP, XMPP, email and REST

Fax Servers

Asterisk
Fax Email Gateway
Elastix Unified
Communications distro supporting FAX and Virtual FAX capabilities
ICTFAX,
is an Open Source Foip Software featuring email to fax , fax to email and web to fax based on freeswitch and ICTCoreCommunication
Framework.
Lintad:
Linux Telephone Answering Device - A Voice and Faxmail Server
Hylafax
ICTDialer Is
an Open Source unified communications autodialer and broadcasting software application supporting voice, sms, fax broadcasting.

Development platforms, protocol stacks

Adhearsion:
High-level, highly productive backend telephony development framework based on Asterisk. Written in Ruby.
H323plus:
Open Source H.323 Protocol Stack following on from the original openH323
IVR
for Skype: Open Source example in C#. No hardware required.
OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems,
OpenMGCP: Open Source MGCP Protocol Stack Developed with C and POSIX APIs,
OpenSS7: SS7 Protocol
Stack
ooh323c:
Open Source H.323 Protocol Stack Developed in C
++Skype C++
library for skype add-on platform independent software development. It is platform
independent, easy to use, and easy to extend because of the flexible library design, inspired by modern C++ design ideas. Performance is one of the goals.

Radius Servers

Aradial:
Radius server and Billing for VoIP
BSDRadius:
Radius server for VoIP
Interlink
RADIUS Server RADIUS Server Software
RadBox RADIUS Server + Billing System. (For a work, you nead instal Framework 2.0)

Billing

See Open
Source Billing Systems
BillRun BillRun
- - Open Source Billing Solution, designed for Big Data

Codecs

See Codec
Software

Middleware

Ernie: Open Source Python based applications platform for VoIP and presence based applications
Mobicents:
The most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.
TALK:
Web based CTI Solution (AJAX client) which provides call control, presence and directorty features.

Suite Solutions

Zoontelecom:
Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)

CTI Dialer utilities

Asterisk phonebook A
common shared phone book directory for Asterisk PBX
TALK Powerful
directory management and scalable architecture to create Click to call or Select and Dial applications + AJAX libraries to implement these features in your web site.

转自：https://www.voip-info.org/wiki/view/Open+Source+VOIP+Software