Tags : Browse Projects

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Project Tox

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  Analyzed 9 days ago

Free and secure distributed Skype replacement.

37.4K lines of code

24 current contributors

24 days since last commit

10 users on Open Hub

Moderate Activity
5.0
 
I Use This

Restcomm

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Claimed by TeleStax Analyzed 15 days ago

The Restcomm Communication Platform is the best architecture to create, deploy and manage services and applications integrating voice, video and data across a range of IP and legacy communications networks. It drives convergence with the following key enablers : Restcomm Connect API, JAIN-SLEE, SIP ... [More] Servlets, SS7 Stack , Resource Adaptors, SMSC and USSD Gateway, Rich Multimedia Services, Presence Services/RCS, Diameter/AAA Services, XMPP Services, Web Services and others. Restcomm is supported by TeleStax [Less]

5.23M lines of code

113 current contributors

15 days since last commit

9 users on Open Hub

Very High Activity
4.375
   
I Use This

Coccinella

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  Analyzed over 1 year ago

Coccinella is a free and open-source cross-platform communication tool with a built-in whiteboard for improved collaboration with other people.

145K lines of code

0 current contributors

over 6 years since last commit

9 users on Open Hub

Activity Not Available
5.0
 
I Use This

WengoPhone

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  Analyzed almost 7 years ago

OpenWengo is a community project focussed on communication over IP, including VoIP, instant messaging and video phonecalls. Its main product is the Wengophone - a standards-based softphone and multi-protocol IM client. The successor of WengoPhone is QuteCom.

109K lines of code

0 current contributors

about 10 years since last commit

9 users on Open Hub

Activity Not Available
4.0
   
I Use This

Ring

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Claimed by Savoir-faire Linux Analyzed 9 days ago

Ring is a robust, standards-compliant enterprise softphone, for desktop and embedded systems. It is designed to handle several hundred calls a day. Ring is available under the GNU GPL license, version 3.

228K lines of code

38 current contributors

12 days since last commit

8 users on Open Hub

Very High Activity
4.4
   
I Use This

A2Billing

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  Analyzed 11 days ago

A2Billing complements the Asterisk project by enabling the following features on both TDM and VoIP calls: Traditional calling card services Credit limit on both pre-paid and post-paid customers Callback services Residential VoIP services Wholesale minutes termination Monthly/weekly free ... [More] calling packages Invoicing Paypal, Moneybookers and Authorize.net integration. The project is easy to use and is frequently seen on FreePBX installations to bring accountability to small offices' phone usage. For ITSP and traditional telco wholesale usage it has been seen to easily scale to millions of minutes per month, with 100,000s rates across many trunks. Work is in progress to further enhance A2Billing's scalability and availability. [Less]

252K lines of code

0 current contributors

about 2 years since last commit

8 users on Open Hub

Inactive
4.0
   
I Use This

CallWeaver

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  Analyzed about 3 years ago

CallWeaver is a community-driven vendor-independent cross-platform open source PBX software project (formerly known as OpenPBX.org). It was originally derived from Asterisk. Now it supports analog and digital PSTN telephony, multi-protocol voice over IP telephony, fax, software-fax, T.38 fax over IP ... [More] and many telephony applications such as IVR, conferencing and callcenter queue management. [Less]

251K lines of code

0 current contributors

over 7 years since last commit

8 users on Open Hub

Activity Not Available
4.5
   
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sipp

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  Analyzed 16 days ago

Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features ... [More] the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. Since 1.1rc4, RTP play (voice and RFC2833 DTMFs) is also supported. [Less]

49.6K lines of code

7 current contributors

4 months since last commit

8 users on Open Hub

Low Activity
5.0
 
I Use This

BombusMod_miss.loveless

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  Analyzed about 3 years ago

bombusmod_edited by miss.loveless

61.1K lines of code

0 current contributors

over 4 years since last commit

7 users on Open Hub

Activity Not Available
5.0
 
I Use This

opus codec

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  Analyzed 23 days 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.

69.9K lines of code

22 current contributors

27 days since last commit

6 users on Open Hub

Moderate Activity
5.0
 
I Use This