Search results
Results From The WOW.Com Content Network
Intertex SIP transparent routers, firewalls and ADSL modems, for broadband deployments and SOHO market; Juniper Networks Netscreen and SRX firewalls include complete SIP Application Layer Gateway support; Linux Netfilter's SIP conntrack helper fully understands SIP and can classify (for QOS) and NAT all related traffic; Netopia Netopia supports ALG
OPAL implements a wide range of Voice over IP (VoIP) protocols for voice, video and fax data over IP networks. It is written in C++ and released under the Mozilla Public Licence. OPAL utilises the PTLib portable library that allows OPAL to run on a variety of platforms including Unix/Linux/BSD, MacOSX, Windows, Windows mobile and embedded systems.
Telepathy is a software framework which can be used to make software for interpersonal communications such as instant messaging, Voice over IP or videoconferencing.Telepathy enables the creation of communications applications using components via the D-Bus inter-process communication mechanism.
SIP is an open source software tool used to connect computer programs or libraries written in C or C++ with the scripting language Python.It is an alternative to SWIG.. SIP was originally developed in 1998 for PyQt — the Python bindings for the Qt GUI toolkit — but is suitable for generating bindings for any C or C++ library.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. [1] SIP is used in Internet telephony , in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE ).
JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality:
Gizmo5 (formerly known as Gizmo Project and SIPphone) was a voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. [ 1 ]