VaxVoIP SIP Server SDK (Software Development Kit) contains sample codes, technical manual, COM component (VaxTeleServerCOM.dll) and demo application. It is a complete package to develop a SIP server.

It is designed in a way that it exports maximum control, so that a vendor/developer could use it with other development tools and could add advanced custom dial-plans, create custom queues (priority, circular, random etc.), easily use it with third party components, connect it to any database servers, use and integrate Win32 APIs, as well as many more applications.

VaxVoIP SIP Server SDK contains single COM (Component Object Model) component VaxTeleServerCOM.dll COM component, and can be used with any application development tool, VC++, C#, VB.NET, Delphi, C++, etc.
VaxVoIP COM components exports certain functions and events, and those functions and events can be used to develop a complete SIP server. Please see TECHNICAL MANUAL for more details.

VaxVoIP SIP Server SDK includes sample code in different development tools. SAMPLE CODES are available on our website. Flow diagrams are also provided to understand the sample codes.

VaxVoIP SIP Server SDK is designed and developed in a way that if you run your VaxVoIP integrated SIP server on a computer with multi-core processor/CPU (core2duo, dual core, quad core, hex core), then VaxVoIP SDK divides its processing load among all cores of the CPU to increase the efficiency and handle more SIP clients.

VaxVoIP SIP Server SDK fully supports multi-processor machines. If you run your VaxVoIP SDK integrated SIP server on a system containing more than one processor, it will utilize the power of all the processors and spread its processing load across all processors used—vastly increasing the efficiency of your IP-Telephony network. This delivers a high gain, and allows your CPU/processor to handle high numbers of SIP clients and calls.

One can easily create queues of any type (priority, circular, random, custom) and can also play music on queue calls. Please see SAMPLE CODES (advance) for more details.

One can easily add ring group functionality that allows to have multiple phones ring when one extension or number is dialed. Please see SAMPLE CODES (advance) for more details.

Call pick-up functionality can be added very easily that allows one to answer someone else's call. For further details, please download SAMPLE CODES (advance) and run it.

It is really easy to develop call parking feature that allows a person to put a call on hold at one extention and continue the conversation from any other telephone set or extention. For further details, please download SAMPLE CODES (advance) and run it.

It is incredibly easy to develop call barging functionality that is used in call-centers for training puposes and allows a call center manager to listen in on live calls without the caller or call center agent knowing (known as silent call monitoring) and then barg into the call to speak with both the agent and the caller (known as call barging). Please see SAMPLE CODES (advance) and DEMO for more details.

VaxVoIP SIP Server SDK supports server side multi-user call conference, in which multiple calls can be added to a single conference. Such feature can be used to create server side chat rooms. Please see SAMPLE CODE and DEMO for more details.

One can easily add call hold feature in the SIP server developed by using VaxVoIP SIP Server SDK. Music can also be played to the on-hold call.

VaxVoIP SIP SDK integrated SIP server can initiate all types of call transfer operations (blind, attend, regular etc) and it can easily process call transfer requests from different SIP clients (softphone, hardphone, ATA, wifi phone, etc.). Please see SAMPLE CODE and DEMO for more details.

One can easily add functionality in VaxVoIP SIP SDK integrated SIP server to record conversations into wave (.wav) files. These features can be used to develop call-center, IVR systems, leave message and many other services to keep the call records.

VaxVoIP SIP Server SDK exports a set of functions to play wave (.wav) files to the call conversations. Buffered based compression technology is introduced in VaxVoIP SIP SDK to save CPU cycles. Under this model, VaxVoIP’s technology compresses the wave data just one time, buffers it, and simply uses that same buffered data every time. While playing the wave data, buffered based compression technology minimizes voice compression processing load on the CPU and increase the server’s efficiency.

VaxVoIP SIP Server SDK exports functions to generate/send DTMF digits. Two types are supported to generate DTMF digits: 1) InBand, and 2) RTP based (RFC 2833). Please see TECHNICAL MANUAL for more details.

During the call conversation, if VaxVoIP SIP SDK receives DTMF digits, then it detect those digits and trigger events.
Three types of DTMF detection are supported: 1) Inband, 2) RTP based (RFC 2833) and 3) SIP based (INFO). These features are used to develop call-center services, IVR systems, and many others. Please see SAMPLE CODE and TECHNICAL MANUAL for further details.

VaxVoIP SIP Server SDK is friendly to those SIP clients who are behind the NAT/firewalls. If any SIP client (softphone, hardphone, ATA) is behind the NAT/firewall, then it can easily be connected to VaxVoIP SDK based SIP server without making any extra settings (stun, estun, port forwarding).

VaxVoIP SIP Server SDK supports SIP client authentication procedures. Under that procedure, SIP clients (softphone, hardphone, ATA, wifi phone) can authenticate and register to the SIP server developed by using VaxVoIP SIP SDK. Please see SAMPLE CODES and TECHNICAL MANUAL for further details.

VaxVoIP SIP Server SDK is compatible to SIP protocol. Any SIP based client (softphone, hardphone, ATA, wifi phone, etc.) can connect and register to a VaxVoIP SDK integrated SIP server, allowing the client to send and receive phone calls.