Saturday, December 18, 2004


Open source VOIP projects

By Klaus Darilion

VoIP bookmarks from Klaus Darilion
Below you will find descriptions and links to SIP and RTP stacks, applications, test utilities, SIP proxies, SIP PBXs and STUN server and clients. Most of them are open source :-), but not all of them :-(
If you have any comments please feel free to contact me: --> klaus.darilion at <-- There are also other VoIP related portals and link collections.
Note: I mainly searched for C/C++ stacks and applications. There also exist a lot of stacks and applications for other programming languages, especially for java. If you are looking for Java stacks/applications, please ask Google (search for: NIST java jain).
RTP Stacks (mainly open source C/C++ stacks)
jrtplib: A very nice, simple C++ RTP stack. Works on Windows, Linux.... ; License: Free; Homepage: This stack is not symmetrical, but you can use my version of rtpconnection.cpp (for jrtp version 2.8) to make it symmetrical. (send RTP and receive RTP on the same port, send RTCP and receive RTCP on the same port).
Common Multimedia Library: from UCL London, includes RTP stack; C; License: Free; Homepage:
ortp: C; License: LGPL; Homepage:; without RTCP, from linphone
GNU ccRTP: C++; License: GPL (with linking exception); Homepage:
LIVE.COM Streaming Media: C++; License: LGPL; Homepage:
Morgan RTP DirectShow Filters: C++; License: ?; Homepage:; based on liveMedia library
RTP from C++; License: VOCAL; Homepage:
RTPlib: RTP library from Lucent Technologies/Cloumbia University; C; License: Non-exklusive source code license; Homepage:
librtp: C; License: GPL; Homepage:; from Gnome-o-phone
Microsoft RTC API: The Mircosoft RTC API is a high level SIP and RTP Stack. It's included in Windows XP and also comes with the several Windows Messenger. Version 1.2 introduced a lot of new features is behaves strange when used with other SIP clients. Developer Homepage:
sipXmediaLib: Part of pingtel's open source releases for VoIP. License: LGPL; Homepage:
SIP Stacksexternal SIP stack comparison
dissipate: C++; Linux, requries the qt-library, License: GPL; Homepage:; The original dissipate by Billy Biggs.
dissipate2: C++; Linux, requries the qt-library, License: GPL; Homepage:; A enhanced dissipate, is part of the kphone distribution.
GNU osip: C; Linux+Windows+...; License: LGPL; Homepage:; Also known as libosip. Note: The interface of osip has been changed and from now on it will be called osip2! Download the tar file from
GNU eXosip: C; Linux+Windows+...; License: GPL; Homepage:; The extensible osip: "...It aims to implement a simple high layer API to control the SIP for sessions establishements and common extensions. Once completed, this eXtended library should provide an API for call management, messaging and presence features.... Download the tar file from
SIP from C++; Linux+Windows+...; License: Vovida Software License; Homepage:
resiprocate: C++; Linux+Windows+...; Includes now a high level API (DialogUsageManager) which supports refers, ... License: VOCAL; Homepage:
Microsoft RTC API: The Mircosoft RTC API is a high level SIP and RTP Stack. It's included in Windows XP and also comes with the several Windows Messenger. Version 1.2 introduced a lot of new features is behaves strange when used with other SIP clients. Developer Homepage:
sipXtackLib: Part of pingtel's open source releases for VoIP. License: LGPL; Homepage: There is also a high level call library (sipXcallLib), which implements JTAPI in C++.
libmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage:
RTP Applications
RAT - Robust Audio Tool; Supports a large number of codecs, ... License: Free; Homepage:
JMF - Java Media Framework: Can receive and send RTP streams; Homepage:
MP3/RTP Plugin for Winamp: Homepage:
Vomit - Voice over Missconfigured Internet Telephones: Plays back captured voice conversation; Homepage:
RTP Tools: Several RTP utilities from the Columbia University; Homepage:
UDP Packet Reflector/Forwarder: A tiny tool which forwards or reflects UDP packets. You can also add delay and packet loss. Very useful if you want to test RTP applications. Homepage: As I was not able to compile this tool I searched and found a binary somewhere in the web. You can download it local
SIP Phones (SIP User Agents)
x-lite, x-pro: A SIP client for Windows; Mac OS and Windows CE, A really nice SIP UA with a lot of features. The light version is free and really rocks, the pro version not. Supports multiple proxies.
eyeP Phone Lite: A SIP client for Windows, a FWD version is available for free
SIPPS: SIP softphone with answering machine and a lot of features. They have also integrated support for for SIP-PSTN termination. A Demo for testing is available. The configuration is a bit weird (what's the difference between a proxy and a redirect server?).
MSN Messenger: Microsofts Messenger, Version 4.6 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of Homepage:; local download of Version 4.6 for Windows NT (2000).
MSN Messenger: Microsofts Messenger, Version 4.7 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of Homepage:; local download of Version 4.7 for Windows XP.
Microsoft portrait: Windows SIP client that supports Audio, Video and IM. Uses RTC API 1.2 and therefore has poor compatibility with other SIP clients.
Ubiquity User Agent: Java based SIP Client for Windows, very useful, you have to register (free) to get an license; Homepage:
EZ-Phone (Evaluation Version): SIP Phone for Windows; Homepage:
MySIP: SIP User Agent from Siemens; Homepage:
SJPhone: SIP and H.323 Softphone for Windows, Linux and PocketPC from: The configuration for SIP is a little bit tweaky. And there must not be another SIP client running on port 5060 or the SJPhone won't work.
Linphone: A SIP Softphone for Linux (GNOME), needs libosip ans oRTP; Homepage:
KPhone: A SIP Softphone for Linux (KDE); Homepage:
Vovida: Complete SIP Suite for Linux (Uaser Agent, Proxy, ...), very, very big software contruct; Homepage:
Siphon: Linux SIP Softphone; Homepage:
ActXPhone: An ActiveX-Control SIP Softphone based on the Microsoft Real Time Communications (RTC) API.
sipXphone: Part of pingtel's open source releases for VoIP. License: LGPL; Homepage: This softphone also requires lots of other libraries from the sipX... software at
Shtoom: An open source, cross plattform SIP client written in Python. License: LGPL; Homepage:
Cornfed SIP-UA: A SIP user agent for Linux. License: Free for non-commercial use (binary distribution); Homepage:
MiniSIP: An open source SIP user agent for Linux which runs on PDAs. It is based on several libraries, including libmsip, a C++ SIP stack. Homepage:
SIP Test Utility
sipsak: SIP Swiss Army Knife, very useful test utility (Linux); Homepage:
SIPNess: Ortena Networks SIP Messenger, very useful test utility for windows; Homepage:
SIP request generator: A web based generator of SIP requests: send SIP requests to SIP UAS and waits for final response: Download at or test it online at Download at
NastysipA simple Linux-program from SX-Design that generates bogus SIP-messages and sends them to any peer. Download at
sipXtest: Part of pingtel's open source releases for VoIP. License: LGPL; Homepage:
SIP Forum Test Framework (SFTF): A Framework to test SIP devices for common errors. License: GPL; Homepage:
callflow: a powerful SIP call flow visualizer; Homepage:
SIP Scenario Generator: a powerful SIP call flow visualizer; Homepage:
SIPp: a powerful SIP performance testing tool sponsered by HP; Homepage:
SIP Applications (Proxy, Location Server)
Sip Express Router (ser): Highspeed GNU SIP proxy with a lot of features and a lot of ongoing development. Homepage: A really cool SIP proxy - I like it! You can also take a look at the development homepage with web CVS. At the beginning you should read the admin guide and the mailing lists archive.
Ser Media Server (sems): Media Server add-on for ser SIP proxy. Homepage: Supports voicemail, IVR, SIP/PSTN gateway ...
Asterisk: Linux Software PBX with Gateway, SIP Proxy, Gateway (SIP, H.323, PSTN, ...); Homepage:
sipd: A Linux SIP proxy from SX-Design written in C (GPL):
partysip: A Linux SIP proxy based on osip2 (LGPL). Developer homepage is at:, you can download tar packages from:
mysip: A SIP proxy server from Siemens for Windows platforms. Homepage:
Fomine RTC server: A SIP proxy server for Windows which uses its own SIP stack (does NOT need the RTC API) Homepage: The unregistered version can be used up to 5 users.
sipXpbx: Part of pingtel's open source releases for VoIP. License: LGPL; Homepage: This PBX combines various sipX applications like a SIP proxy (sipXregistry, sipXproxy), a media server (sipXvxml) and lots more.
yate: Yet Another Telephony Engine - a PSTN gateway. License: GPL; Homepage: This gateway supports H.323, SIP and zaptel (->asterisk) based PSTN cards.
STUN server and clients
mystun: STUN server and client library from the guys. License: GPL, Homepage: You have to download the file via CVS.
Vovida STUN server: STUN server and client library/application for Linux and Windows from the Vovida guys. License: Vovida Software License 1.0, Homepage: The files are hosted at sourceforge.
NAT traversal ALG (application level gateway)This applications can be installed on a linux NAT-box. They will rewrite your SIP messages and have some kind of UDP/RTP proxy for the media stream.
SaRP - SIP and RTP proxy: Perl implementation, License: GPL, Homepage:
siproxd: Siproxd is a proxy/masquerading daemon for the SIP protocol based on osip. License: GPL; Homepage:

The VOIP - a reference guide to all things VOIP This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. ''Welcome to! Please e let me know at Thanks. ^ News This section is for news, ie news reports, press releases, product release announcements etc.
Research: Peer-to-Peer Internet Telephony using SIP PDF
Iconv application module for character conversion.
Version 0.9.2 of LDAPget application module released. bugfix.
Over 5 million VoIP subscribers worldwide - DMeurope Story
Interviews with bkw, twisted and David Mandelstam
Interview With drumkilla, the manager of the stable branch of Asterisk
Interview With bkw, one of the developers of Asterisk
Land Grab: What If Wal-Mart Got in the WiMax/VOIP Business? Robert X. Cringely Editorial
Interview With Mark Spencer, the creator of Asterisk - Slashdot Entry Cordless VoIP card DECT now compatible with SKYPE, MSN Messenger and any Audio applications based on Windows. Works in dual mode with SIP and Audio.
More News Information
Asterisk: Open Source PBX
SIP Express Router: An Open Source SIP proxy/router
VOIP Tools
Analog Telephone Adapters: VoIP analog telephone adapters ATA
Digital Telephone Adapters: VoIP Digital/TDM telephone adapters
VOIP Phones: VoIP phones both hardware and software
VOIP Gateways: VOIP to PSTN gateways (also known as: Media Gateways)
VOIP PBX and Servers
VOIP Routers
USB Phone
VOIP Session and Border Controllers and NAT traversal solutions
Open Source Software
Commercial Software
VOIP Billing
ENUM - map E164 telephone number into VoIP addresses
FXS-FXO Converters
Dial Pulse to Touchtone DTMF Converters
VOIP Test Equipment: How to analyze the speech quality of VoIP equipment
VOIP Silicon Chips specifically designed to support VOIP
VOIP Service Providers: VSPs, the next generation telco
Wireless VOIP
Low Bandwidth VOIP Using VOIP on dial-up or other low speed connections
VOIP over Satellite Using VOIP over satellite connections
QoS - Quality of Service
VOIP Codecs
VOIP Bandwidth Requirements
How To Debug and Troubleshoot VOIP
VOIP sites: VOIP sites on the Internet
VoIP Policy State and Federal VoIP policy and regualatory issues
VoIP Training: Seminars, tutorials, on-line classes
IP Protocols: SIP, LTP, H.323, SCCP, MGCP, Megaco, IAX, STUN, ENUM, TRIP, SIMPLE, RTP, PINT, SCTP, T.37, T.38, COPS
ITU protocols: SS7, ISUP
ITU related standards: P.1010
OSP - Open Settlement Protocol
Markup Languages
IVR Presentation and dialog management: VoiceXML
Call control / conferencing / call routing: CCXML
IVR / Speech recognition definition: SRGS
IVR / Speech synthesis definition: SSML
IVR / prompting / recording / conferencing / DTMF / Voice: CallXML
Standards Organizations: IETF, ITU, CCITT, ANSI, ISO, IEC, ETSI, IEEE, W3C
VoIP User Groups: List of local VoIP User Groups
Industry: SIP Forum, H.323 Forum, SIPfoundry, ChinaVoIP
Traditional Telephone Network
Analog Telephone Information
PBX features
Telecom terms: ANI, DNIS, PBX, CENTREX, E1, T1, BRI, PRI, ISDN, DID, Channelbank, Numbering plans, NEBS, CALEA, LATA, 911, PSAP, V and H Coordinates, NPA-NXX, Telecom Dictionary, CLEC, LNP, more
Telco Engineering Information Suggestions and Questions
How to add information to this wiki
Suggestions: Put your requests and suggestions here
Search VOIP sites on the Internet

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