Projects tagged ‘sip’ and ‘telephony’

Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris. It provides all of the features you would expect from a PBX and more as it enables ... [More] developers to build customized voice applications of many types. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. [Less]

Robust Open Source (GPL) SIP (RFC3261) server implementation. Among features: VoIP SIP registrar/proxy/router/application server, TLS secure communication, AAA, ENUM, LCR, load balancing, NAT ... [More] traversal, OSP, CPL, SNMP, SIMPLE IM&Presence, DNS failover. The source code for development is hosted now in GIT at http://sip-router.org [Less]

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 ... [More] digital PSTN telephony, multi-protocol voice over IP telephony, fax, software-fax, T.38 fax over IP and many telephony applications such as IVR, conferencing and callcenter queue management. [Less]

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 ... [More] , video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. [Less]

ZOIPER 2.0 Free SIP and IAX is a user-friendly softphone, compatible with Asterisk or any other SIP or IAX capable system. Great VoIP calling experience with features like: SIP + IAX / IAX 2 ... [More] protocols Available codecs – GSM, ulaw, alaw, speex, ilbc STUN support STUN server per account Two accounts DTMF tones sending Echo cancellation Codec settings per account Account password encryption Call history Hold function Quick dial panel Optional Automatic pop-up window for incoming call Call logs Minimize on tray Minimize on start up Always on top Adaptive Jitter Buffer Support for multiple audio devices Address book Quickdial pad Automatic user registration Call transfer Voice mail message information Portable ZoIPer with portable devices (like USB, flashcards, etc.) Multilanguage support [Less]

OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server ... [More] , is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS 'unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design. What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class. [Less]

sipXecs IP PBX (Enterprise Communications Server) is an open source software implementation of a Session Initiation Protocol (SIP) based unified communications system (IP PBX). Similar to a ... [More] traditional private branch exchange (PBX), it allows a number of attached telephones to make calls to one another, and to connect to other telephone services including the PSTN and SIP Trunking services. sipXecs IP PBX is an open source alternative to commercial PBX offerings. It is easy to use and scales to several thousand users. [Less]

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 ... [More] can also act as a generic media relay as well as gateway RTP sessions between IPv4 and IPv6 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]

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 ... [More] Callback services Residential VoIP services Wholesale minutes termination Monthly/weekly free 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]

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. ... [More] It also reads XML scenario files describing any performance testing configuration. It features 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]