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Restcomm

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Claimed by TeleStax Analyzed 4 months 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 2 years since last commit

10 users on Open Hub

Activity Not Available
4.375
   
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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
   
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Mostly written in language not available
Licenses: gpl3

opus codec

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  No analysis available

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.

0 lines of code

7 current contributors

0 since last commit

9 users on Open Hub

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

openss7

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  Analyzed about 16 hours ago

openss7 STREAMS and protocol stacks

3.4M lines of code

1 current contributors

about 4 years since last commit

9 users on Open Hub

Inactive
0.0
 
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sipp

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  Analyzed about 22 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]

49.7K lines of code

6 current contributors

2 days since last commit

8 users on Open Hub

Moderate Activity
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]

667K lines of code

53 current contributors

2 days since last commit

7 users on Open Hub

High Activity
5.0
 
I Use This

Sofia-SIP

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  Analyzed 4 months 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]

333K lines of code

0 current contributors

about 6 years since last commit

6 users on Open Hub

Activity Not Available
5.0
 
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rtpproxy

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  Analyzed about 6 hours 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]

113K lines of code

4 current contributors

about 2 months since last commit

6 users on Open Hub

Very Low Activity
5.0
 
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pjsip

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  No analysis available

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.

0 lines of code

3 current contributors

0 since last commit

6 users on Open Hub

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

CDR-Stats

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  Analyzed about 1 hour ago

CDR-Stats is a free and open source call detail record analysis and reporting software for Freeswitch, Asterisk and other types of VoIP Switch. It allows you to interrogate CDR to provide reports and statistics via a simple to use powerful web interface. It is based on the Django Python ... [More] Framework, Celery, SocketIO, Gevent and MongoDB. Star2Billing S.L. is the company behind the development of CDR-Stats. [Less]

53.9K lines of code

0 current contributors

over 9 years since last commit

6 users on Open Hub

Inactive
5.0
 
I Use This