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Open Source VOIP applications, both clients and servers (开源sip server & sip client 和开发库)

SIP Proxies

  • 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 to Kamailio
  • 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:

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

IAX clients

TURN servers and RTP Proxies

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

XCAP servers

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

IVR platforms

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)

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 andICTCore Communication 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

Codecs

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.

Here’s a list of open source VoIP software available on the market:

SIP proxies

  • Clearwater: An open source IP Multimedia Subsystem implementation for vastly scalable utilisation in the Cloud. Its SIP routing components are built onPJSIP.
  • Elastix: An open source unified communications server that supports chat, mail, fax and video conferencing.
  • JAIN-SIP Proxy: Can be used in SIP proxy servers, SIP user agents and test tools, as well as to build session border controllers, resource adapters for JAVA, and SIP servlet implementations.
  • Kamailio: Formerly OpenSER, this is a SIP server and registrar with TLS support for VoIP and real-time communications. It also offers SIP authentication, diameter, RADIUS, ENUM and least-cost-routing. Other features include instant messaging, load balancing, fail-over routing, accounting, and support for backend systems.
  • MiniSIPServer: A small Perl POE-based SIP proxy that offers enterprise communication services like ringing- or hunting-group, follow-me and call queue. Can run on multiple operating systems and virtual machine systems. It also works on an IPv6 network.
  • MjServer: Cross-platform SIP proxy, registrar and redirect server. It’s written in JAVA and based on the MjSIP stack, which is used to initiate voice/video/multimedia sessions for interactive and noninteractive applications. Also includes call control, user agent and session border controller implementations.
  • Mumble: An open source, low-latency voice chat software mainly for gaming use. It also offers encrypted communication and key authentication. You can even recognise friends across servers.
  • MySIPSwitch: SIP proxy server that allows the use of multiple SIP accounts with just one SIP login. It also allows central management of any SIP-based VoIP service.
  • NethidPro3.0.6: An open source SIP encryption bridge – an anti-blocking VoIP encryption system that supports using HT serial encryption VoIP gateway.
  • Net-SIP: A Perl SIP framework which comes with a stateless proxy.
  • OpenJSIP: An open source distributed standalone SIP proxy, registrar and location service that is run by JAVA VM. It’s based on NIST SIP (various SIP applications, tools and libraries in JAVA) and comes from the JAIN-SIP proxy.
  • OpenSBC: SIP proxy, registrar and B2BUA that includes NAT traversal, ENUM, and reference implementation of session border controller. It can be expanded and modified based on personal needs.
  • OfficeSIP: An open source SIP server for Windows to make voice and video calls and deliver instant messages between computers for free.
  • OpenSIPS: Derived from OpenSER, it’s a SIP proxy/server for voice, video, presence, IM, and other SIP extensions.
  • party SIP: A flexible and customisable SIP proxy server with different plugins to add or remove capabilities. Users can disable useless functions and allow new ones with little development.
  • Repro: From the reSIProcate project, it’s a free open source SIP server. It comes with a full implementation of federated VoIP and a built-in web user interface for fast setup.
  • Remwave: A high performance cross-platform SIP proxy that’s written in JAVA.
  • SaRP: SIP proxy in Perl, but a cross-platform C++ version is in the works.
  • SIP Express Router (SER): High performance SIP proxy, router and registrar. It also offers SIP authentication, diameter, RADIUS and ENUM. It can easily fix operational issues such as broken network components and is scalable from small office environments to acting as a PBX or Asterisk replacement.
  • Sippy B2BUA: SIP back-to-back user agent server software. It’s a call controller, maintaining complete call state and participating in all call requests. It can perform accurate call accounting, fail-over call routing, prepaid rating and billing, and more.
  • Siproxd: A masquerading SIP proxy server that can run on Linux, Mac OS X, and other Unix derivatives.
  • SIPVicious: A suite of tools for auditing SIP-based VoIP systems.
  • SipXecs: A complete, native SIP PBX solution for Linux. It provides instant messaging, voicemail, user portals, admin GUI, and plug & play management.
  • Vocal VoIP Software: Has a SIP server with network stack. Can be customised to business needs and also provides call routing, call control and billing information for Linux. It’s capable of adaptation and scalability.
  • Yxa: A set of SIP server applications written in the Erlang programming language.

Monitoring software

  • Aheeva Contact Centre Suite: An all-in-one contact centre that offers remote monitoring, contact management and network analytics.
  • Homer: An open source SIP capture system and monitoring application. It comes with HEP, IP Proto 4 (IPIP) encapsulation and port mirroring/monitoring.
  • Kamailio: It offers SNMP (Simple Network Management Protocol) monitoring, which monitors network devices for conditions that require administrative attention.
  • MonAst – The Asterisk Monitor: HTML interface that acts as an operator panel for Asterisk for displaying user/peer status and calls. It uses a reverse AJAX, Python and PHP for originating, transferring and hanging up calls, as well as managing queues and meet me rooms.
  • nProbe: An open source NetFlow Probe for gigabit networks. Offers precise VoIP traffic monitoring to build accurate analysis applications.
  • ntop: Provides VoIP network traffic monitoring with simple metrics.
  • OrecX: Records, monitors and replays calls for contact centres and business VoIP providers. The software can be localised and customised for free.
  • PJSIP-perf:  Monitors and tracks SIP transaction and call performance.
  • SIP-CallerID: Retrieves and looks up SIP caller ID.
  • SIP Inspector: Written in JAVA, this software monitors incoming and outgoing SIP messages, and much more.
  • VoIPmonitor: An open source network packet sniffer for archiving, monitoring and troubleshooting quality of SIP VoIP calls.

Stacks and libraries

  • eXosip: An extended oSIP library which makes it easier to use the SIP protocol for multimedia session establishment for either VoIP or multiplayer gaming.
  • IvrWorx: VoIP prototyping and testing framework – high-level LuaInterface to SIP/RTSP/MRCP in order to test complicated VoIP networks.
  • Libre: Portable and generic library for real-time communications with a complete SIP stack with IPv4/v6 support.
  • MjSIP: JAVA-based SIP library for J2SE and J2ME platforms.
  • MSRP Library: Message Session Relay Protocol peer library and its relay extension written in Python.
  • NIST SIP 1.2: Series of SIP libraries, applications and tools in JAVA.
  • OpalVoIP: The Open Phone Abstraction Library is a C++ multiplatform, multiprotocol library for fax, video and voice over IP and other networks.
  • Open SIP Stack: SIP stack that includes ENUM, Presence (XMPP/SIMPLE) and NAT traversal. It comes with a platform agnostic stack implementation of RFC3261 so development can be done in various platforms, including Windows and Linux.
  • oSIP Library: Provides multimedia and telecom software developers an interface for initiating and controlling SIP-based sessions in their applications.
  • PJSIP: High performance SIP protocol stack written in C and with language binding for Python. It’s also customisable, portable and has a small footprint.
  • reSIProcate: SIP stack and sample application from SIPfoundry that works in various operating systems like Unix and Windows. Can be used for implementing proxies, instant messaging and gateways.
  • SailFin: Adds SIP support to the JAVA GlassFish application server.
  • SIP.js: JavaScript library for WebRTC and SIP signalling.
  • Sofia-SIP: A SIP user agent library and SIP stack implementation that includes support for STUN and presence.
  • Twisted: Protocol stacks and applications for Python with SIP support.
  • Vovida SIP Stack: An implementation of the SIP protocol for initiating voice calls over IP networks.
  • YASS: Stateful SIP stack used inYate and written in C++ that can be used for a proxy/server in a single or multi-thread model.
  • Yxa: A transaction stateful SIP stack written in Erlang.

Platforms

  • Aheeva Contact Centre Suite: An open source PBX system for IP telephony. It offers call centre features such as call routing, contact management functions and research functions.
  • Asterisk PBX: An open source telephony engine and toolkit for building communications applications, such as IP-PBX and IVR systems, VoIP gateways and conference bridges. Also includes voicemail, call queuing and automated call distribution.
  • Bayonne:  A GNU project IVR server for users to easily integrate with telephony.
  • Elastix: Unified communications software that supports IP-PBX and IVR capabilities for IP telephony.
  • Evolution PBX: Asterisk-based PBX system that makes it easier and more cost effective to integrate existing phone systems with new VoIP systems.
  • FreePBX EcoSystem: An open source PBX platform for building robust and powerful communications solutions for businesses and call centres. Can be customised and adapted to your changing needs and run onsite or in the cloud.
  • Freeside CRM: Open source platform for VoIP, ISPs, hosted solutions, online businesses and service providers, with billing, CRM, automation and trouble-ticketing capabilities.
  • FreeSWITCH: An open source telephony platform for voice calls and chat. Can be used as a PBX system, media gateway and media server for hosting IVR applications. Also features voicemail, conferencing, recording, and more.
  • ICTDialer: An open source auto-dialler software that includes graphical IVR designer tools. It also supports voice, SMS and fax broadcasting.
  • jPBXLite: JAVA-based VoIP (SIP) PBX system that supports voicemail, voice conferences, call queues and an IVR system.
  • Kamailio: It’s also a high end, open source PBX system that supports instant messaging and presence.
  • Open PBX: PBX software platform for small offices and large call centres. Features include voicemail, auto-attendant and automatic call distribution. It can also be customised and extended with its highly compact Perl code.
  • OpenVXI: An IVR platform that implements VoiceXML, which is used to make IVR applications in PBX solutions.
  • OrecX: Available in both open source and open platform formats for recording, monitoring and replaying calls.
  • PBX4Linux: Software-based ISDN PBX platform with H.323 gateway for Linux.
  • SEMS: Free open source SIP express media server that has IVR capabilities.
  • SIP Witch: Pure SIP PBX call and registration server.
  • SipXecs: An open source SIP PBX for Linux with built-in IVR (voicemail and auto-attendant). The platform also comes with instant messaging, presence, and FreeSWITCH-based conferencing.
  • Sipwise sip:provider CE: A SIP-based VoIP soft-switch platform that can be used to build a variety of VoIP business models/systems, which include voice/video calls, conferencing, presence, voicemail and instant messaging.
  • snom ONE: Formerly pbxnsip, it includes IP-PBX and IVR platforms, along with a fax server, unified messaging, conferencing, outbound dialling, etc.
  • Switchvox: IP-PBX platform with call recording and visual voicemail.
  • Yate: Telephony engine that supports SIP and H.323. It offers open source PBX/PABX and IVR platforms, instant messaging, voicemail, VoIP, conferencing and call centre service.

References:

04-26 23:00
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