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VOCAL
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VOCAL software

Vocal and Vovida are founders of SIPfoundry.org

Applications at Vovida.org


VOCAL – Vovida Open Communications Applications Library

The lates update to VOCAL was in April of 2003, software release 1.5.0.

VOCAL provides the development community with software and tools needed to build new and exciting VoIP features, applications and services. The software in VOCAL includes a SIP based Redirect Server, Feature Server, Provisioning Server, Policy Server and Marshal Proxy along with protocol translators from SIP to H.323 and SIP to MGCP. Our hope is that these modules will act as building blocks to help you create better, faster and stronger VoIP systems.

B2BUA
The Back-To-Back User Agent (B2BUA) is a Session Initiation Protocol (SIP) based logical entity that can receive and process INVITE messages as a SIP User Agent Server (UAS). It also acts as a SIP User Agent Client (UAC) that determines how the request should be answered and how to initiate outbound calls. Unlike a SIP proxy server, the B2BUA maintains complete call state and participates in all call requests.

Load Balancer Proxy
The Load Balancer is a very simple proxy that is useful in SIP-based VoIP installations where there are multiple ingress proxy servers. The Load Balancer permits pooling these servers, thereby eliminating the need to balance user demands for connectivity through a complicated provisioning algorithm.

Open G.729(A) Initiative
VoiceAge, a Montreal-based provider of voice compression and communications technologies, announces the “Open G.729(A) Initiative”: a developer’s version of the G.729(A) codec software to be used for product development and non-commercial purposes.

Open AMR Initiative
VoiceAge, a Montreal-based provider of voice compression and communications technologies, announces the “Open AMR Initiative”: a developer’s version of the AMR codec software to be used for product development and non-commercial purposes.

OpenOSP
OpenOSP is an open source implementation of a server for the Open Settlement Protocol (OSP). It was developed jointly by Cisco Systems, Inc. and Data Connection Limited.

Ramalho G.711 Lossless Compression (RGL) Codec
The Ramalho G.711 Lossless Compression (RGL) codec is an open source implementation that was generously contributed by a member of the community.

SIP Residential Gateway
The SIP Residential Gateway (SIPRG) is an open source application based on the Session Initiation Protocol (SIP). The SIPRG is an IP Telephony Gateway that allows a SIP User Agent to make and receive calls between the Public Switched Telephone Network (PSTN) and a SIP-based network such as VOCAL. This software was generously contributed by Tata Infotech Ltd of India.

SIPSet
SIPSet is a SIP User Agent with a GUI front end that works with the Vovida SIP stack. You can use the SIPSet as a soft phone, to make and receives phone calls from your Linux PC.

SIPTiger
SIPTiger is a web-based provisioning utility for Cisco’s line of 7960 Session Initiation Protocol (SIP) IP phones and Cisco SIP proxy servers (CSPS). This utility is useful for anyone deploying Cisco 7960 SIP IP phones.

STUN Server
The STUN (Simple Traversal UDP (User Datagram Protocol) through NATs (Network Address Traversal)) server is an implementation of the STUN protocol that enables STUN functionality in SIP-based systems. The STUN server tar ball also include a client API to enable STUN functionality in SIP endpoints.

WinRTP
WinRTP is a COM component which can originate RTP (Realtime Transport Protocol) media from a microphone and terminate RTP media on a speaker.


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