SIP Express Router (SER) is a very fast and flexible SIP (RFC3621) proxy server. Written entirely in C, SER can handle thousands calls per second even on low-budget hardware. A C Shell like scripting language provides full control over the server's behavior.
It's modular architecture allows only required functionality to be loaded. Currently the following modules are available: digest authentication, CPL scripts, instant messaging, MySQL support, a presence agent, radius authentication, record routing, an SMS Gateway, a jabber gateway, a transaction module, registrar and user location.

Serweb is a web based provisioning interface of The SIP Express Router. It is provided as is without any warranty under the terms of GPL. To install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web. Then, change config.php to reflect your local configuration. Serweb is written in PHP.

This package is based on the package 'speex' from project 'openSUSE:Factory'.

Speex is a patent free audio codec designed especially for voice
(unlike Vorbis which targets general audio) signals and providing good
narrowband and wideband quality. This project aims to be complementary
to the Vorbis codec.

The STUN (Simple Traversal of UDP through NATs (Network Address Translation)) 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. In addition there is a command line UNIX client and a graphical windows client that check what type of NAT the user is using.

STUN is an application-layer protocol that can determine the public IP and nature of a NAT device that sits between the STUN client and STUN server.

Twinkle is a SIP-based soft phone for making telephone calls over IP networks.

Vale is a library for handling media streams over IP.

Libwww is a general-purpose Web API written in C for Unix and Windows (Win32). With a highly extensible and layered API, it can accommodate many different types of applications including clients, robots, etc. The purpose of libwww is to provide a highly optimized HTTP sample implementation as well as other Internet protocols and to serve as a testbed for protocol experiments. Libwww also supports HTTPS, through OpenSSL.

The call processing subsystem provides the core to all sipX end points
or user agents (e.g. sipXphone and sipXvxml. The sipXcallLib library
contains technologies related to call processing:

* cp - An abstracted Call Processing model

* ptapi - A C++ version of JTAPI call model

* ps - An abstraction for virtual (or physical) phone set objects
(e.g. lamps, hookswitch)

* tao - A transport layer designed to decouple (and remote) the
application layer from call processing.

This package contains the neccessary header files and static libraries
helpful to build projects from the sipX library

This project is used to hold code used in common by a number of the sipX family servers. It is not intended to be useful as a standalone project.

Configuring one phone can be a daunting task, configuring a hundred
phones can be maddening. sipXconfig leverages Jetty, Axis
SOAP, postgresql and other JEE technologies to get the job done.

This library provides a set of adapters for using a media subsystem. Three
adapters currently have been implemented. They are for Pintel media processing,
GIPS VoiceEngine, and GIPS ConferenceEngine. The applications must select an
appropriate library during the link time in order to use the right media
subsystem.

The sipXmediaLib includes all of the audio processing used in the
sipXphone and sipXvxml projects. For example, the library contains
audio bridges, audio splitters, echo supression, tone generation
(e.g. DTMF), streaming support, RTCP, G711 codecs, etc.

The sipXpbx project is a comprehensive small/medium scale Enterprise SIP PBX. It combines:

* Call routing sipXproxy
* sipXregistry registry/redirect server,
* the subscribe/notify framework and message waiting indication package from sipXpublisher,
* Media Server sipXvxml with auto-attendant and voice mail applications,
* PBX and phone configuration support from sipXconfig.

This library provides a set of classes that provide an operating
system abstraction from a majority of OS provided functions. All of
the sipX projects use this library to ensure easy porting to any
operating system. The library currently provides classes that
encapsulate functions and operations for:

* Threads
* Locks and Mutexes
* Semaphores
* Messages and Queues
* Timers
* Time and Date
* Sockets
* File and Directory
* Operating System Processes
* Dynamic loading of shared libraries and symbols

Two RFC 3261 compliant SIP proxies:
* sipForkingProxy
* sipAuthProxy
These proxies are used in the sipXpbx project. However they may be used
independently from the PBX as stand-alone SIP proxies. They may be used
separately or together. The sipXregistry may also be used with the forking
proxy. Though it is not necessary. The forking proxy provides service level
routing as well as parallel and serial forking. The auth proxy provides AAA
services.

A modular server for handling SIP event subscriptions; event package types
can be added through a dynamically linked library interface, configured through
a simple XML plugin configuration file.

sipXregistry is the registry/redirect server component of the sipX family.
It shares a number of common components with the rest of the family, but
can be used as a standalone server.

This project provides an RFC 3261, 3263 complient SIP stack. This
stack is built upon a modest foot print HTTP stack. Both the SIP and
HTTP stack are intended to be usable in embedded systems. However they
are fully functional (i.e. there where no compromises made to in
functionality to achieve the ability to embed the stacks.). The
primary interface to the SIP stack it through the SipUserAgent
class. This project depends upon the sipXportLib OS abstraction layer.

Voice XML processing engine.
With scripts to implement the voicemail and auto-attendant functions of a PBX.