Tags : Browse Projects

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QuteCom (formerly WengoPhone)

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  Analyzed over 2 years ago

QuteCom is the new name for the open source softphone previously known as WengoPhone, a standards-based softphone and multi-protocol IM client. It is a community project focussed on communication over IP, including VoIP, instant messaging and video phonecalls.

2.92M lines of code

0 current contributors

over 4 years since last commit

6 users on Open Hub

Activity Not Available
3.83333
   
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pjsip

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  Analyzed about 1 year ago

pjsip is a professionally supported open source comprehensive multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.

599K lines of code

4 current contributors

about 1 year since last commit

6 users on Open Hub

Activity Not Available
5.0
 
I Use This
Licenses: GPL-2.0+, pjsip Sta...

opus codec

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  Analyzed about 17 hours ago

The Opus codec is designed for interactive speech and audio transmission over the Internet. It is designed by the IETF Codec Working Group and incorporates technology from Skype's SILK codec and Xiph.Org's CELT codec.

68.5K lines of code

19 current contributors

22 days since last commit

6 users on Open Hub

Moderate Activity
5.0
 
I Use This

The Open Source Enterprise Telephony Recording and Retrieval System

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  Analyzed over 1 year ago

Oreka is an enterprise telephony recording and retrieval system with a Web-based user interface. It supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP, and audio sound devices, and runs on multiple operating systems and database systems. It can record audio from most PBX and ... [More] telephony systems, such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc. It is being used in call centers and contact centers for quality monitoring (QM) purposes. [Less]

53.7K lines of code

4 current contributors

over 1 year since last commit

5 users on Open Hub

Activity Not Available
5.0
 
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openSpeak

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  Analyzed over 6 years ago

openSpeak is a VoIP solution currently in development. It’s basically aimed at clan or casual gamers who like to chat with their friends/teammates while playing. There is currently support for Windows and Linux, other platforms might be added later on.

8.19K lines of code

0 current contributors

over 8 years since last commit

5 users on Open Hub

Activity Not Available
5.0
 
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Sofia-SIP

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  Analyzed about 12 hours ago

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform for Sofia-SIP is ... [More] GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL. Actively maintained by FreeSWITCH (https://freeswitch.org/fisheye/changelog/freeswitch/libs/sofia-sip). [Less]

332K lines of code

0 current contributors

about 6 years since last commit

5 users on Open Hub

Inactive
5.0
 
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FusionPBX

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  Analyzed 1 day ago

FusionPBX is an open source project that provides a customizable and flexible web interface to the very powerful and highly scalable multi-platform voice switch called FreeSWITCH. It can be run on the operating system you are comfortable with and hardware of your choice. Unlimited extensions ... [More] , voicemail-to-email, music on hold, call parking, analog lines or high density T1/E1 circuits and many other features. It provides the functionality your business needs and brings corporate level phone system features to small, medium and large businesses. [Less]

387K lines of code

40 current contributors

2 days since last commit

5 users on Open Hub

Very High Activity
5.0
 
I Use This

Adhearsion

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  Analyzed about 18 hours ago

Adhearsion is an open-source voice application development framework written in Ruby. Adhearsion users write applications atop the framework with native Ruby syntax and a simplified Domain-Specific Language for call management enabling users to call into their code. Adhearsion rests above a ... [More] lower-level telephony platform, namely Asterisk, and provides a framework for integrating with various resources, such as SQL, LDAP and XMPP (Jabber). Adhearsion has... *An elegant dialplan system for writing the code which controls a live phone call *A sophisticated Asterisk Manager Interface library *An events subsystem *A reuseable component architecture *Ability to re-use existing Ruby on Rails database models with ActiveRecord/ActiveLDAP *Easy interactive communication via XMPP instant messages [Less]

64.3K lines of code

0 current contributors

about 1 year since last commit

5 users on Open Hub

Very Low Activity
4.2
   
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trixbox CE

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  No analysis available

trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)

0 lines of code

0 current contributors

0 since last commit

5 users on Open Hub

Activity Not Available
0.0
 
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Mostly written in language not available
Licenses: GPL-2.0+

ice4j

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  Analyzed over 1 year ago

A Java implementation of the STUN and TURN and ICE protocols. This project is a continuation of the Stun4J initiative. The goal of the project is to provide VoIP Java developers with a means of handling NATs and firewalls. The project is a joint effort of the SIP Communicator and TurnServer open ... [More] source projects but is entirely based on standards and can be used with any other VoIP client or STUN/TURN server. [Less]

32.3K lines of code

2 current contributors

about 2 years since last commit

4 users on Open Hub

Activity Not Available
5.0
 
I Use This