Tags : Browse Projects

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KMobileTools (KDE)

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Claimed by KDE No analysis available

KMobileTools is an OpenSource project (GPL2 license) that aims to be a complete management system for mobile phones. It's written in C++ using Qt and KDE libraries. It's using KParts technology, so it can be embedd in every application (like in Kontact), it can read and write KDE AddressBook, full ... [More] import/export, read and write SMS, export them in KMail, and much more. All this depending on your phone capabilities, of course. CALL FOR DEVELOPERS: please check on the kde-apps project page how you can help and contribute. Without a new developer team KMobileTools won't be finished in near future. HELP IS REQUIRED! [Less]

0 lines of code

0 current contributors

0 since last commit

11 users on Open Hub

Activity Not Available
4.75
   
I Use This
Mostly written in language not available
Licenses: gpl

openss7

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  Analyzed 1 day ago

openss7 STREAMS and protocol stacks

3.4M lines of code

1 current contributors

over 5 years since last commit

11 users on Open Hub

Inactive
0.0
 
I Use This

oFono

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  No analysis available

0 lines of code

22 current contributors

0 since last commit

11 users on Open Hub

Activity Not Available
5.0
 
I Use This
Mostly written in language not available
Licenses: gpl

Homer SIP Capture

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  Analyzed about 16 hours ago

HOMER is a robust, carrier-grade, scalable SIP Capture system and Monitoring Application with HEP/HEP2, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box, ready to process & store insane amounts of signaling with instant search, end-to-end analysis and ... [More] drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling [Less]

62.2K lines of code

2 current contributors

about 1 month since last commit

10 users on Open Hub

Low Activity
5.0
 
I Use This

A2Billing

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  No analysis available

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]

0 lines of code

0 current contributors

0 since last commit

10 users on Open Hub

Activity Not Available
4.0
   
I Use This
Mostly written in language not available
Licenses: gpl

Yate

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  No analysis available

Yet Another Telephony Engine is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine ... [More] , maximizing communications efficiency and minimizing infrastructure costs for businesses. [Less]

0 lines of code

5 current contributors

0 since last commit

10 users on Open Hub

Activity Not Available
4.25
   
I Use This
Mostly written in language not available
Licenses: gpl

Jami

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Claimed by Savoir-faire Linux Analyzed about 17 hours ago

Jami is the only software required for peer-to-peer (without a server) communication that respects the freedom and privacy of its users. Jami is the simplest and easiest way to connect with people (and devices) with instant messaging, audio, and video calls over the Internet and LAN/WAN ... [More] intranets. Jami is a free/libre, end-to-end encrypted, and private communication platform. Jami calls are directly between users, as it does not use servers to handle calls. This gives the greatest privacy, as the distributed nature of Jami means all calls are only between participants. One-to-one and group conversations with Jami are enhanced with instant messaging, audio and video calling, recording and sending audio and video messages, file transfers, screen sharing, and location sharing. [Less]

890K lines of code

27 current contributors

3 days since last commit

9 users on Open Hub

High Activity
4.4
   
I Use This

sipp

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  Analyzed about 19 hours 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]

50.1K lines of code

6 current contributors

20 days since last commit

8 users on Open Hub

Moderate Activity
5.0
 
I Use This

jSS7- Java SS7 Protocol Stack / Framework

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  Analyzed 1 day ago

JSS7 is an implementation of SS7 telephony protocol in Java, aims to create an open source, multiplatform, SS7 protocol stack. This project is inspired by openss7 (www.openss7.org) and asterisk (www.asterisk.org)

452K lines of code

0 current contributors

over 1 year since last commit

7 users on Open Hub

Very Low Activity
0.0
 
I Use This

rtpproxy

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  Analyzed 1 day ago

The Sippy RTPproxy is a high-performance software proxy for RTP streams that can work together with SIP Express Router (SER), OpenSER or Sippy B2BUA. Originally created for handling NAT scenarious it can also act as a generic media relay as well as gateway RTP sessions between IPv4 and IPv6 ... [More] networks. RTPproxy was developed by Maxim Sobolev and now is being actively maintained by the Sippy Software, Inc. The RTPproxy supports some advanced features, such as remote control mode, allowing building scalable distributed SIP VoIP networks. The nathelper module included into the SIP Express Router (SER) or OpenSER as well Sippy B2BUA allow using multiple RTPproxy instances running on remote machines for fault-tolerance and load-balancing purposes. [Less]

276K lines of code

4 current contributors

2 months since last commit

6 users on Open Hub

Moderate Activity
5.0
 
I Use This