sip

sip-redirect is a tiny SIP redirect server. sip-redirect supports IPv4 and IPv6, but the IPv6 support is optional.

The RFC 3261 was the base for this simple and very configurable implementation. There is neither TCP nor multicast support programmed in.

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.

Sofia-SIP project 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 GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.

Here are some key features of "Sofia SIP":

SIP features

· Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.

SIP Offer-Answer module

· Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.

NAT traversal support

· Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.

SIP security support

· Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.

Brekeke SIP Server registers and authenticates users, and routes calls between user agents. The product has original NAT traversal functionality as well as flexible control routing functions.

Supported operating systems include Microsoft Windows XP/2000, Solaris 10, and Mac OS X. Free for Personal and Academic Institution use.

Benefits

· Multiple OS Support
· Intuitive Web-based Administration Tool
· Support for Media Stream over NAT Traversal
· Multiple-Domain Support
· Highly Interoperable with third party products and services
· Customizable to meet unique needs through Dial Plans & Plug-ins
· Flexible Scalability

Here are some key features of "Brekeke SIP Server":

· Registrar Service Brekeke SIP Server receives REGISTER requests from SIP UAs and updates its database appropriately. Using the registrar function, you can receive calls from any SIP UA using your unique SIP-URI.
· Call Routing Brekeke SIP Server will route SIP requests from a SIP UA, or other server, to the most appropriate SIP-URI address based on its registrar database and Dial Plan. Brekeke SIP Server supports SIP Redirect feature which allow servers to redirect a request back to SIP UA.
· NAT Traversal Brekeke SIP Server enables SIP UAs behind the NAT to talk with other SIP UAs, including video over NAT traversal. Using NAT-enabled firewalls ensures the security that you want to retain, while giving users the ability to make media calls over the internet between different networks.
· Dial Plan With a Dial Plan you can use regular expressions to define matching or filtering rules for headers and IP addresses in the SIP packets. Brekeke SIP Server’s Dial Plan increases compatibility between SIP compliant products and provides added capability and flexibility for creating complex call routing.
· Upper/Thru Registration Upper/Thru Registration allows for easy configuration of parallel users of preexisting or other SIP servers. This feature allows establishing SIP communication through ITSP lines or third-party SIP servers.
· Authentication By specifying authentication settings on REGISTER or INVITE requests, you can limit calls that go through Brekeke SIP Server. Authentication Plug-in is available for the users who wish to use an existing user directory service.
· Multiple-Domain Hosting Brekeke SIP Server can host multiple domains on one server install. This feature allows user to manage multiple domains under one server setup.
· Session Management The real-time session management is available through GUI. View session status or manually terminate the active calls.

SIP SDK for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.

The conaito SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo suppression, noise suppression, reverb suppression and mute microphon

Key Features

* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec)
* Encrypt SIP account settings (protect your SIP account settings in websites)
* Secure Weblicensing (protect your license in websites)
* Multi-User conference support
* Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* Call Hold support
* Call Transfer support
* Instant text messaging (MIME) support and typing indication
* Mute microphone/speaker for each line
* DNS SRV resolution for SIP servers (RFC 3263)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* PLC (Packet Lost Concealment)
* AGC, VAD, AEC, Noise Concealment

...and much more. Try it today!

Sip Zip is a file compression tool much similar to the popular Winzip. This little application has all the basic features an archiver must have.

Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.

This software contains a high performance VoIP conferencing client capable of delivering crystal clear sound for both low and high-bandwidth users and SIP compatible devices (hardware and software).

It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller , acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here are some key features of "VoIP SIP SDK":

· Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider.
· VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, iLBC Codec)
· Registration on SIP Server (SIP Registrar).
· Instant text messaging.
· Microphone and Speaker Visualization support.
· Microphone and Speaker Volume with Mute support.
· Audio device selection.
· Fully-customizable user interface.
· Packetloss resistant (by using iLBC codec).
· Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
· Works with all kind of Internet connections.
· Royalty free licensing
· No Yearly/Monthly fee
· Very easy to incorporate
· AGC (auto gain controller).
· Acoustic echo cancellation or suppression.
· Noise cancellation or suppression.
· Reverb cancellation or suppression.
· VAD (Voice activity detection).


Requirements:

· Visual Basic .NET
· Visual C++ .NET
· Visual C# .NET
· Delphi .NET
· Visual Basic
· Visual C++
· Delphi
· Visual Basic 6.0, all development environments with a ActiveX support


Limitations:

· 30 days trial period

One of the features of Python that makes it so powerful is the ability to take existing libraries, written in C or C++, and make them available as Python extension modules. Such extension modules are often called bindings for the library.

SIP is a tool that makes it very easy to create Python bindings for C and C++ libraries. Python-SIP was originally developed to create PyQt, the Python bindings for the Qt toolkit, but can be used to create bindings for any C or C++ library.

SIP comprises a code generator and a Python module. The code generator processes a set of specification files and generates C or C++ code which is then compiled to create the bindings extension module. The SIP Python module provides support functions to the automatically generated code.

The specification files contains a description of the interface of the C or C++ library, i.e. the classes, methods, functions and variables. The format of a specification file is almost identical to a C or C++ header file, so much so that the easiest way of creating a specification file is to edit the corresponding header file.

SIP makes it easy to exploit existing C or C++ libraries in a productive interpretive programming environment. SIP also makes it easy to take a Python application (maybe a prototype) and selectively implement parts of the application (maybe for performance reasons) in C or C++.

Whats New in This Release:

· This release adds support for consolidated and composite modules.
· It adds support for pickling classes and enums.

SIP Phone DLL allows to make PC-PC, PC-phone, or phone-phone calls or create Instant Messaging (IM) sessions over the Internet VOIP SIP Phone SDK brings protocol support for ActiveX. With this SDK one can create a simple program to connect and start speaking with anyone with a direct IP Address (provided no NAT Router or Firewall is set) or use the Gateway or the Gatekeeper to connect to /or from PSTN lines. Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex, our VOIP Implementation Team at Research-Lab will guide you remotely for the same. VOIP SIP DLL Soft Phone SDK brings SIP protocol support for ActiveX. With this SDK one can create in minutes a VOIP phone program to connect and start speaking with anyone with a direct IP Address (provided no NAT Router or Firewall is set) or use the Gateway or the Gatekeeper to connect to /or from PSTN lines. Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex, to tackle this our VOIP Implementation Team at Research-Lab will guide you remotely for the same.

This software will help you connect and start speaking with anyone with a direct IP Address or use the Gatekeeper or the Gateway to establish a connection to and from PSTN lines.

Here are some key features of "SIP Phone DLL":

· Make and answer phone calls
· Detect tone and pulse digit from the phone line
· Capture Caller ID
· Support blind transfer
· Single Step transfer/conference
· consultation transfer/conference
· hold, unhold
· Control of the local phone handset
· microphone and speaker of the modem
· Send and receive faxes
· Play and record on the phone line or sound card
· Play music in background mode
· Silence detection
· VU Meter
· Voice recognition and voice synthesis support
· Full control over the serial port device ZModem file transfer utility (with encryption and decryption)

SIP Express Router (SER) is a high-performance, configurable, free Voice over IP/SIP ( RFC3261 ) server.

It can act as registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, voice mail, offline message service, user call preferences (CPL) etc.

Web-based user provisioning, serweb, available.

Cornfed SIP User Agent is a Session Initiation Protocol (SIP) based softphone for your IBM-compatible Personal Computer running the Linux operating system.

The Cornfed SIP User Agent allows you to make Internet phone calls using an Advanced Linux Sound Architecture (ALSA) or Open Sound System (OSS) sound card with speakers and microphone as your telephone handset.

Here are some key features of "Cornfed SIP User Agent":

· Supports SIP (RFC 3261), SDP (RFC 2327), and RTP (RFCs 3550 and 3551)
· Automated detection of Residential Gateways using Network Address Translation (NAT)
· Supports Digest authentications for registrations and outbound INVITEs
· Support for loose proxy routing using Record-Route and Route headers
· Handles forking of outbound INVITEs by proxies
· Supports re-INVITEs for changes to media transport
· Supports G.711 mu-Law and a-Law voice codecs
· Supports RFC 2833 DTMF tone generation
· Supports SIP compact header forms
· Gnome GUI and CLI clients
· Multi-threaded implementation

The Cornfed SIP User Agent is provided free of charge for personal use for users of the Linux operating system. The program is provided as a binary distribution only. The Cornfed SIP User Agent is specifically designed with embedded and mobile wireless devices in mind.

Development is under way to bring this client to other platforms. If you are interested in licensing this technology for your commercial application, please contact Cornfed Systems at 410-404-8790.

Whats New in This Release:

· Minor bug fixes.
· Changed readline functionality to used raw input rather than cooked mode.
· Fixed bugs with display of status messages that caused client to hang.
· Disabled removal of registration on client exit.

Telepati SIP Phone Freeware description
Freeware VOIP SIP Soft Phone. Tested and works with 75% of the providers Telepati SIP Phone Freeware allows you to make PC-PC phone-phone calls over the Internet. Developed using Research Labs VOIP SIP Phone SDK, this free soft phone brings SIP protocol support for ActiveX.

With this phone once you set the gatekeeper proxy with the username and password from your providers, you can connect and start speaking with anyone on the internet. Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines.

Though this implementation might turn out fairly complex, our VOIP Implementation Team at Research-Lab will guide you remotely for the same. Tested and works with 75% of the providers.

Whats New in This Release:

Connection Timeout Value Increased for Increasing the number of supported Service Providers Range

SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server .

SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.

Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.
SERs configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.

Here are some key features of "SIP Express Router":

· accounting
· digest authentication
· CPL scripts
· ENUM support
· instant messaging
· MySQL support
· PostgreSQL support
· a presence agent
· Radius authentication and accounting
· Diameter authentication
· record routing
· SMS gateway
· Jabber gateway
· NAT traversal support transaction module
· registrar
· user location

SER has been extensively and successfuly tested with many SIP products from other vendors (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, and many others). It has been powering our SIP services continuously for more than two years.

VaxVoIP SIP SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based IP-Telephony make and receive phone calls feature in your web pages and software applications. It accelerates the development of SIP based soft phone with your own GUI (graphical user interface) and brand name.

It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering.

In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK.

In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK. Hands-free or Internet telephony imposes several problems. The principal one is due to the coupling between loudspeaker and microphone. The loudspeaker signal is echoed back to the microphone and transmitted back to its origin. As a result the far-end participant perceives this as an echo.

VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and provides high quality of output speech.

Jitter buffers are used to smooth delay variations in received audio by buffering the packets and adjusting their rendering. The result is a smoother delivery of audio to the user.

Packet Loss Concealment (PLC) is a technique used to mask the effects of lost or discarded packets. PLC is generally effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates.

User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT/firewall.