Changes between Initial Version and Version 1 of Projects_Using_PJSIP

Timestamp:

May 22, 2012 4:16:38 AM (12 years ago)

Author:

bennylp

Comment:

--

Projects_Using_PJSIP

@@ +1 @@
+{{{#!table style="border: dash;"
+  {{{#!td align=left valign=top style="border: dash"
+  Title
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+  Description
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://greenj.org Greenj]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+"GreenJ is an open source Voice-over-IP phone software using pjsip and Qt. It can easily be used to build your own VoIP phone system. Our approach was not to build a complete phone with user interface, but instead provide an application that handles only the communication. The program logic and user interface are separated from the application by using an integrated browser. We use webkit as browser engine, which is well integrated into Qt (QWebView). A Javascript interface handles all communications between application and webpage. This means that you can use GreenJ as it is and create your VoIP phone entirely in HTML and !JavaScript."
+
+[[Image(http://www.loremipsum.at/wp-content/uploads/2011/09/GreenJ.jpg)]]
+
+Author: Lorem Ipsum Mediengesellschaft m.b.H.
+
+Added: 2011/11/18
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://microsip.org.ua/ MicroSIP]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+"MicroSIP - free portable SIP softphone based on PJSIP stack for Windows OS, allowing high quality VoIP calls (p2p or on regular telephones) via open SIP protocol." 
+
+Goals of project:
+ * 100% work with any number of NATs in both sides
+ * Voice quality
+ * Small RAM usage
+ * Usability
+
+Screenshots:
+
+[[Image(http://microsip.org.ua/images/microsip-1.png)]][[Image(http://microsip.org.ua/images/microsip-2.png, 35%)]][[Image(http://microsip.org.ua/images/microsip-3.png)]]
+
+Author: Dmitry Valegov
+
+Added: 2011/08/18
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://code.google.com/p/csipsimple/ csipsimple]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+A native SIP client for Android. Screenshots:
+
+
+[[Image(http://www.r3gis.fr/blog/public/device4.png, 35%)]]
+
+Author: Régis Montoya
+
+Added: 2010/09/15
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://sourceforge.net/projects/pjsip-jni/ pjsip-jni]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+A Java Native Interface (JNI) wrapper for pjsip, supporting PJSUA API.
+
+
+Author: Florian Hackenberger
+
+Added: 2010/09/15
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://www.tlphn.com/ Telephone]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+Telephone is a softphone for Mac that integrates with Mac OS X address book. Screenshots:
+
+
+[[Image(http://telephone.googlecode.com/svn/site/account.png)]]
+[[Image(http://telephone.googlecode.com/svn/site/call.png)]]
+[[Image(http://telephone.googlecode.com/svn/site/address-book-card.png)]]
+[[Image(http://telephone.googlecode.com/svn/site/incoming-call-notification.png)]]
+
+Author: Alexei Kuznetsov
+
+Added: 2010/04/20
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://artemisa.sourceforge.net/ Artemisa]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+'''TransferHTTP: A SIP Integrated Web Browser for HTTP Session Mobility and Multimedia Services'''
+
+
+Artemisa is a VoIP/SIP-specific honeypot software designed to connect to a VoIP enterprise domain as a user-agent backend in order to detect malicious activity at an early stage. Moreover, the honeypot can play a role in the real-time adjustment of the security policies of the enterprise domain where it is deployed.
+
+
+Artemisa uses the Python module provided by PJSIP
+
+[[Image(http://artemisa.sourceforge.net/images/deployment.jpg)]]
+
+Author: Rodrigo do Carmo
+
+Added: 2010/04/20
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://transferhttp.mozdev.org/ TransferHTTP]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+'''TransferHTTP: A SIP Integrated Web Browser for HTTP Session Mobility and Multimedia Services'''
+
+
+Web session migration is one of the ways of improving the web browsing experience. Other ways include the use of bookmarks and web history synchronization. This extension, TransferHTTP, provides an HTTP Session Mobility and Multimedia Services using SIP.
+
+[[Image(http://transferhttp.mozdev.org/implementation_framework.gif)]]
+
+Author: Michael Adeyeye
+
+Added: 2010/03/16
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://code.google.com/p/opensoftphone/ opensoftphone]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+This SIP softphone is written in Java as an eclipse RCP application. It uses the pjsip SIP stack for connecting to SIP servers. The phone runs on Windows and Linux. It would run on Mac OS too, but manually compiling it is necessary because of the JNI bindings to pjsip. The Java-JNI binding which are used by the phone are hosted on sourceforge.net, but are currently included in the SVN tree.
+
+
+Author: Florian Hackenberger
+
+Added: 2010/01/25
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://http://infrahip.hiit.fi/ Host Identity Protocol for Linux (HIPL)]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+The '''[http://tools.ietf.org/html/draft-ietf-hip-nat-traversal Host Identity Protocol (HIP)]''' and the related architecture form a proposal to change the TCP/IP stack to securely support mobility and multi-homing. Additionally, they provide for enhanced security and privacy and advanced network concepts, such as moving networks and mobile ad hoc networks. The InfraHIP project studies application related aspects of HIP, including APIs, rendezvous service, operating system security, multiple end-points within a single host, process migration, and issues related to enterprise-level solutions.
+
+
+Author: Miika Komu.
+
+[[Image(http://freshmeat.net/screenshots/c8/01/c8016a07abc9d2f5b5c4fd7300115e96_medium.jpg?1237057621)]]
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://github.com/imankulov/network-emulator/ Media Impairments Simulator]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+Network-emulator is a simple utility intended to test how network losses affects speech quality in VoIP-based applications. Experimenter can set up loss rate, bandwidth, encoder options and select one of the packet loss suppression algorithm.
+
+
+Emulator can help quickly obtain these measures:
+
+ * compare encoding quality for different codecs and codecs modes.
+ * estimate the impact of the loss level and distribution on the speech quality.
+ * estimate the impact of the different PLC algorithms on the speech quality.
+
+
+
+Author: Roman Imankulov.
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://android.wooyd.org/ VoiDroid (VoIP client for Android)]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+Add VoIP SIP client functionality to Android phones.
+
+
+Author: Jurij Smakov.
+
+[[Image(http://android.wooyd.org/images/voidroid.png, 35%)]]
+
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[/contrib/pjsiptcl/ TCL Wrapper for PJSUA-API]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+See '''[/contrib/pjsiptcl/README.txt README.txt]'''
+
+
+Authors: Antonio F. Cano Damas and Mats Bengtsson.
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://svsip.free.fr SvSIP]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+'''[http://svsip.free.fr SvSIP]'''
+is a project to port PJSIP on Nintendo DS (and also iPod Touch it
+seems!). Please check it out, it's cool!
+
+[[Image(http://svsip.free.fr/local/cache-vignettes/L180xH135/kbd-min-openmoko-c09af.png)]]
+
+Author: Samuel Vinson.
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://code.google.com/p/sipeksdk/ Sipek SDK]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+.. "SipekSDK is a small VoIP Software Development
+Kit written in C#. The goal of SipekSDK is to offer simple and easy to
+use API for VoIP developers."
+
+
+Author: Sasa Coh
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[https://sites.google.com/site/sipekvoip/ SIPekPhone]'''   
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+.. "Sipek is a SIP phone & messaging
+client based on generic VoIP engine powered by pjsip.org SIP stack.
+Combining voice calls, Instant Messaging and presence in an intuitive
+user interface, Sipek takes you into the world of Voice over IP. The
+project is based on SipekSDK VoIP library. Currently it supports a C#
+wrapper to connect to pjSIP stack. The wrapper (pjsipdll) part of Sipek
+can be used in other .Net projects including windows mobile."
+
+[[Image(http://sipekphone.googlepages.com/Sipek.jpg/Sipek-medium.jpg)]] [[Image(http://sipekphone.googlepages.com/Accounts3.jpg/Accounts3-medium.jpg)]]
+
+Author: Sasa Coh
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://developer.berlios.de/projects/dtmfbox/ dtmfbox]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+.. "The dtmfbox is a tool which can be used to
+control different tasks over telephone keyboard (DTMF). Mostly, it was
+made to run on the AVM FRITZ!Box. "
+
+
+Author: Marco Zissen
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://www.ipcom.at/index.php?id=560/ QjSimple]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+.. "QjSimple is a prototype implementation of a
+cross-platform SIP Client. It is based on the pjsip SIP stack and the
+Qt GUI toolkit. QjSimple can be seen as developer tool and supports the
+following features:
+
+
+ * cross-plattform Windows/Linux 
+ * SIP over UDP/TCP/TLS 
+ * RTP/SRTP 
+ * Instant Messaging 
+ * Presence (SIMPLE)" 
+
+[[Image(http://www.ipcom.at/uploads/pics/qjsimple.gif)]]
+
+Author: Klaus Darilion
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://puppylinux.org/wikka/PuppySIP PuppySip/PSIP]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+Psip is a very simple Voice Over Internet Protocol (VOIP) application based on PJSUA. The main benefit of this application is it small size, around 600k.
+
+
+Author: tmxxine
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''REMWAVE Inc.'s Mac Communicator'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+..''"SIP 2.0 Based Softphone for Mac OSX.
+Integrated with your address book for phone numbers and IM addresses
+(Jabber support to be added soone). Place high quality, cheap phone
+calls over your Internet connection!"''
+
+
+Author: REMWAVE Inc
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://sourceforge.net/projects/aoip/ Audio over IP Interoperability Engine]'''
+
+EBU N/ACIP Reference Implementation
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+This is the reference implementation of European
+Broadcasting Union ('''[http://www.ebu.ch/ EBU]''')'s
+Audio over IP ('''[http://www.ebu-acip.org N/ACIP]''')
+standard.
+
+''"This project aims to build a
+software reference implementation of the EBU standard for the
+transmission of high quality, low latency, audio streams over IP
+networks (EBU-tech 3326)"''.
+
+
+Authors: '''[http://www.bbc.co.uk/rd/index.shtml BBC R&D]''',
+'''[http://www.irt.de/ IRT]'''
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://sourceforge.net/projects/voipforvw/ VoIP for Virtual Worlds]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+The project goal is to <i>"develop
+Open-Source VoIP stack to allow voice communication within Virtual
+Worlds. Specifically as a replacement for proprietary voice chat used
+in !SecondLife"</i>.
+
+
+Author: '''[http://www.3di.jp 3di.jp Inc.]'''
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://code.google.com/p/siphon/ Siphon]'''
+
+VoIP for iPhone and iTouch!
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+The title says it all! Here are some screenshots:
+
+[[Image(http://siphon.googlecode.com/svn/images/SpringBoard.sml.png)]][[Image(http://siphon.googlecode.com/svn/images/Settings_Siphon.JPG,35%)]][[Image(http://siphon.googlecode.com/svn/images/Dialpad.sml.png)]]
+
+Author: Samuel Vinson
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://www1.cs.columbia.edu/%7Esalman/peer/ OpenVoIP]'''
+
+Open Peer-to-Peer VoIP and IM System
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+OpenVoIP is ''"an open source peer-to-peer
+VoIP and IM system of ~1000 nodes running on ~300 !PlanetLab machines.
+OpenVoIP runs Peer-to-Peer Protocol (P2PP) which can be used to
+implement well-known DHTs or unstructured protocols. Unlike OpenDHT,
+where it was only possible to put/get the data, we allow non-!PlanetLab
+nodes to become part of our overlay"''.
+
+
+The OpenVoIP project uses STUN, TURN, and ICE
+features in '''[http://www.pjsip.org/pjnath/docs/html/ PJNATH]'''
+for its NAT traversal.
+
+
+Authors: Salman Baset et all of Columbia University
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://sipsimpleclient.com/ SIP SIMPLE Client]'''
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+SIP SIMPLE client is Python software library built
+on top of PJSIP that together with middleware allows for easy
+development of Internet communications end-points based on SIP and
+related protocols for voice, rich presence, instant messaging (IM) and
+file transfers. Other session types can be easily added by using an
+extensible API.
+
+
+Author: '''[http://www.ag-projects.com AG Projects]'''
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://www.sflphone.org SFLPhone]'''
+
+SIP/IAX Softphone for Linux
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+SFLphone is a SIP/IAX2 compatible softphone for
+Linux. The SFLphone project's goal is to create a robust
+enterprise-class desktop phone. While it can serve home users very
+well, it is designed with a hundred-calls-a-day receptionist in mind.
+
+[[Image(http://sflphone.org/sites/default/files/imagecache/screenshot_preview/multiple-calls.png)]]
+
+Author: '''[http://www.savoirfairelinux.com/ Savoir-faire Linux]'''
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://rtpmobile.sitesled.com/ RTP .NET]'''
+
+Media components for .NET
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+This component allows mobile devices to stream
+voice from Windows Mobile based devices.
+
+
+Author: '''[http://rtpmobile.sitesled.com/Contact.html Anass Kartit]'''
+
+  }}}
+  |-----------------------------------------------------------------------------
+  {{{#!td align=left valign=top style="border: dash"
+'''[http://dev.sipdoc.net/projects/yass/wiki YASS - Yet Another SIP Softphone]'''
+
+SIP softphone, also a simple and small SDK to develop VoIP applications in Python.
+
+  }}}
+  {{{#!td align=left valign=top style="border: dash"
+
+YASS began as a university project, and has been released to the public. Apart from being a SIP softphone, YASS pretends to be a simple and small SDK to develop VoIP applications in Python.
+
+It's based on PJSIP's pjsua Python bindings for the core and the Qt4 libraries for the GUI part. Communication between the core and the GUI is made through callbacks, so it's completely detached.
+
+
+
+Author: '''[http://dev.sipdoc.net/account/show/5 Saúl Ibarra]'''
+
+  }}}
+}}}