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  • sip (254)

    Asterisk-Java

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      Analyzed 27 days ago

    Asterisk-Java, a free Java library for Asterisk PBX integration, consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and ... [More] the Manager API. The FastAGI implementation supports all commands currently available from Asterisk. The Manager API implementation supports receiving events from the Asterisk server (e.g. call progess, registered peers, channel state) and sending actions to Asterisk (e.g. originate call, agent login/logoff, start/stop voice recording). [Less]

    0 lines of code

    11 current contributors

    0 since last commit

    14 users on Open Hub

    Activity Not Available
    4.875
       
    0 Reviews
    I Use This
    Mostly written in language not available
    Licenses: apache_2
    Tags asterisk iax java library sip voip

    Linphone

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      Analyzed 28 days ago

    Linphone is an open source Voice Over IP phone (or SIP phone) that makes possible to communicate freely with people over the internet, with voice, video, and text instant messaging.

    548K lines of code

    25 current contributors

    2 months since last commit

    11 users on Open Hub

    Very High Activity
    3.6
       
    0 Reviews
    I Use This
    Mostly written in C++
    Licenses: gpl
    Tags alsa android blackberry conference encryption h263 iphone j2me linux oss sip softphone 7 more...

    Homer SIP Capture

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      Analyzed 27 days 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

    3 months since last commit

    10 users on Open Hub

    Low Activity
    5.0
     
    1 Review
    I Use This
    Mostly written in TypeScript
    Licenses: gpl3
    Tags callflow capture carrier cluster codecs communication database freeswitch gplv3 hep ipip kamailio 18 more...

    Restcomm

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    Claimed by TeleStax Analyzed 27 days ago

    Restcomm is the only full Stack Communications Platform as a Service (cPaaS). It enables you 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 ... [More] : Communications API Layer and Visual Designer in Restcomm Connect, WebRTC SDKs for Web and Mobile, SMSC, USSD Gateway, GMLC for GeoLocation. It also offers middleware telecom infrastructure with JAIN-SLEE, SIP Servlets, SS7 Stack, SIP Stack, Diameter Stack and SMPP Stack Restcomm is supported by TeleStax [Less]

    5.54M lines of code

    20 current contributors

    about 1 year since last commit

    10 users on Open Hub

    Very Low Activity
    4.375
       
    1 Review
    I Use This
    Mostly written in Java
    Licenses: AGPL3_or_...
    Tags application_server audio cloud communications conferencing cPaaS diameter ims jain-slee java media mediaserver 8 more...

    A2Billing

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      Analyzed 27 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]

    0 lines of code

    0 current contributors

    0 since last commit

    10 users on Open Hub

    Activity Not Available
    4.0
       
    1 Review
    I Use This
    Mostly written in language not available
    Licenses: gpl
    Tags asterisk billing calling card communication h323 iax invoicing ivr linux pbx php 6 more...

    Yate

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      Analyzed 27 days ago

    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]

    388K lines of code

    5 current contributors

    over 2 years since last commit

    10 users on Open Hub

    Inactive
    4.25
       
    0 Reviews
    I Use This
    Mostly written in C++
    Licenses: gpl
    Tags c++ conferencing h323 iax isdn jabber jingle pbx sip softphone ss7 telephony 4 more...

    Jami

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    Claimed by Savoir-faire Linux No analysis available

    Jami (formerly 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.

    0 lines of code

    27 current contributors

    0 since last commit

    9 users on Open Hub

    Activity Not Available
    4.4
       
    0 Reviews
    I Use This
    Mostly written in language not available
    Licenses: gpl3
    Tags c++ c++1y conferencing cross-platform dbus dht gpl gtk+ ice linux messenger secure 8 more...

    sipp

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      Analyzed 28 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]

    49.9K lines of code

    6 current contributors

    2 months since last commit

    8 users on Open Hub

    Low Activity
    5.0
     
    0 Reviews
    I Use This
    Mostly written in C++
    Licenses: gpl
    Tags benchmark c++ dtlf dtmf networking pcap performance programming rtp scenario security sip 8 more...

    FusionPBX

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      Analyzed 27 days 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]

    700K lines of code

    53 current contributors

    2 months since last commit

    7 users on Open Hub

    High Activity
    5.0
     
    0 Reviews
    I Use This
    Mostly written in PHP
    Licenses: mozilla_p...
    Tags callcenter conference freeswitch freeswitchgui fusionpbx pbx php rtp sip voip

    rtpproxy

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      Analyzed 28 days 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]

    119K lines of code

    4 current contributors

    3 months since last commit

    6 users on Open Hub

    Very Low Activity
    5.0
     
    0 Reviews
    I Use This
    Mostly written in C
    Licenses: BSD-2-Clause
    Tags kamailio nat opensips proxy reSIProcate rtcp rtp sdp ser sip sippy tcp 4 more...