The VaxVoIP SIP Server SDK (Software Development Kit) empowers software vendors and service providers to develop their own robust and feature-rich SIP (Session Initiation Protocol) based servers. Moreover, the VaxVoIP SIP Server SDK provides a comprehensive suite of services, including, but not limited to:
Auto-Dialer systems
IVR systems
PC-to-PC call services
Call center systems
Calling card services
VaxVoIP SIP Server SDK is fully compatible with the SIP IP-Telephony protocol and works smoothly with a wide range of SIP-based devices, including softphones, hardphones, Wi-Fi phones, ATA (Analog Telephony Adapters), and other VoIP-enabled SIP devices.
VERSATILE SOLUTIONS FOR BUILDING ADVANCED VOIP NETWORKS
VaxVoIP SIP Server SDK introduces a flexible and innovative solution that integrates a host of advanced capabilities, empowering users to establish their own VoIP network on the Microsoft Windows platform.
VaxVoIP SIP Server SDK incorporates COM (Component Object Model) components and is compatible with a wide array of software development tools and languages, including Visual C++, cSharp, VB.NET, Delphi, Python, Borland C++, and more.
AI-POWERED VOIP SOLUTIONS WITH VAXVOIP SIP SERVER SDK
The SIP Server SDK provides a powerful feature for real-time access to audio PCM data for specific VoIP calls. This capability allows developers to integrate AI-driven technology and machine learning into VoIP-based IP PBX systems, enhancing user experiences with advanced VoIP and AI capabilities.
WHY CHOOSE VAXVOIP AUDIO PCM ACCESS?
Advancing AI Capabilities in VoIP Systems:
Collecting and preprocessing raw PCM audio, selecting the right model, and training it are key to advancing AI capabilities. Integrating machine learning into VoIP and AI into IP-PBX systems enhances call management and communication efficiency. Experience AI-driven VoIP technology and AI-enhanced VoIP communication with advanced integration, resulting in intelligent VoIP solutions that provide superior accuracy and real-time insights.
Integrating AI VoIP Functionalities with VaxVoIP SIP Server SDK:
With the audio data access feature of the VaxVoIP SIP Server SDK, integrate AI VoIP functionalities such as automated call transcription, advanced call routing, intelligent voicemail systems, and dynamic noise suppression. This feature also supports the development of real-time language translation in VoIP PBX systems, enhancing global communication. Experience advanced AI-enhanced VoIP communication through sophisticated AI integration.
Revolutionizing AI VoIP Systems with Innovative SIP Server SDK:
Leverage our innovative SIP Server SDK to revolutionize your AI VoIP systems. Enhance customer service, boost efficiency, and outpace the competition with AI-based VoIP applications. Unlock the full potential of artificial intelligence in VoIP today.ESSENTIAL SOLUTIONS FOR DIVERSE INDUSTRIES AND SMOOTH INTEGRATION
VaxVoIP SIP Server SDK is invaluable for various industries and applications, including:
Banking
Remote PBX for multi-offices
Calling card companies
PBX extensions
Our versatile solution supports a wide range of sectors, ensuring smooth communication and efficiency.
VaxVoIP SIP Server SDK offers the capability to effortlessly interconnect with other SIP-based IP-Telephony gateways and gatekeepers, allowing for smooth deployment within your existing telephony network.
FEATURES
VaxVoIP SIP Server SDK includes advance VoIP (Voice Over Internet Protocol) and IP-Telephony features. It is specially designed for Microsoft Windows operating systems, and works on all Microsoft Windows operating systems.
Highly customizable Solution
The VaxVoIP SIP Server SDK (Software Development Kit) offers a comprehensive package, including sample codes, a technical manual, the COM component (VaxTeleServerCOM.dll), and a demo application. This all-in-one solution empowers developers to create SIP servers with ease.
Designed to provide maximum control and flexibility, the SDK seamlessly integrates with various development tools. It enables developers to craft advanced custom dial-plans, establish custom queues (priority, circular, random, etc.), utilize third-party components effortlessly, connect to diverse database servers, leverage Win32 APIs, and explore numerous other applications.
COM (Component Object Model) based technology
The VaxVoIP SIP Server SDK includes a single COM (Component Object Model) component, VaxTeleServerCOM.dll, which is compatible with a wide range of application development tools, including Visual C++, cSharp, VB.NET, Delphi, Python, Borland C++.
VaxVoIP COM components offer a range of functions and events that can be harnessed for the development of a comprehensive SIP server. For detailed information, please refer to the TECHNICAL MANUAL.
AI MACHINE LEARNING AND MODEL TRAINING
The VaxVoIP Server SDK enables integrated applications to access raw PCM audio data from specific VoIP calls. This real-time audio data access is essential for training machine learning models, as it allows developers to use call audio data to build and refine AI-driven features in AI-based IP-PBX systems, Please see the TECHNICAL MANUAL.
This process of collecting and utilizing real-time audio data from VoIP calls supports the creation of innovative AI-based VoIP applications and artificial intelligence in VoIP. By leveraging this data, developers can drive advancements in communication technology and deliver superior user experiences through the development of cutting-edge AI-based IP-PBX features.
Audio Data Collection for AI Learning:
Collect raw PCM audio data from a variety of sources to create a comprehensive dataset. Ensure the data represents a wide range of scenarios and conditions to facilitate robust learning and improve the performance of AI models.
AI Model and Preprocessing Audio Data:
Before training, raw PCM audio must be preprocessed. This includes resampling to a consistent rate, normalization of volume, segmentation into manageable chunks, feature extraction like MFCCs or spectrograms, and noise reduction to filter out background noise. These steps ensure the audio data is clean and suitable for effective model training.
Selecting the Optimal Machine Learning Model:
Choose an appropriate machine learning model for audio analysis based on the specific application. For tasks such as speech recognition or sentiment analysis, commonly used models include neural networks and recurrent neural networks. Selecting the optimal model is critical for achieving precise and effective results.
Training of AI Model with Preprocessed Audio Data:
Utilize preprocessed PCM audio data to train the selected model. This involves feeding the model the prepared audio features so it can learn patterns and make predictions based on the input data. Proper training ensures the model can effectively understand and process the audio information.
Enhancing AI Capabilities with PCM Data:
Utilize PCM audio data to boost AI capabilities by providing rich, detailed audio input. PCM data supports advanced applications such as speech recognition, sentiment analysis, and noise reduction, enabling models to learn from high-quality, raw audio. This enhances the accuracy and effectiveness of AI-driven solutions by ensuring they have access to precise and unprocessed audio information.
AI-POWERED CALL MONITORING FEATURES FOR VOIP
Accessing the audio PCM data of a call in real time facilitates the development of advanced AI-driven call monitoring capabilities. This technology enables AI to oversee and evaluate each call, providing real-time feedback on call quality and agent performance.
Integrate these advanced AI-powered features into your VoIP infrastructure to transform call management, elevate service quality, and achieve excellence in customer interactions.
Real-Time Performance Feedback:
Receive immediate insights into agent performance, with actionable feedback to drive improvement and maintain high standards of service.
Compliance Monitoring:
Ensure adherence to industry regulations and internal quality standards through AI-powered oversight, reducing risks and maintaining consistency.
Continuous Improvement:
Utilize data-driven recommendations to enhance agent skills, streamline call handling processes, and achieve superior customer satisfaction.
AI-Driven Call Quality Assessment:
Leverage artificial intelligence to monitor and analyse call quality metrics, including clarity, latency, and connectivity issues, ensuring optimal performance.
Natural Language Processing (NLP):
NLP models enable real-time translation of spoken language, facilitating global communication without language barriers.
Intelligent Analytics:
Access comprehensive reports and analytics generated by AI to identify trends, pinpoint areas for growth, and make informed decisions for ongoing optimization.
Easy to understand sample source codes
The VaxVoIP SIP Server SDK offers a diverse collection of sample code snippets optimized for a variety of development tools. These valuable SAMPLE CODES are readily available on our website, allowing developers to gain a comprehensive understanding of our SDK and unlock its full potential for their VoIP projects.
Multi-core processors support
The VaxVoIP SIP Server SDK is meticulously designed and developed to leverage the capabilities of modern computing systems. When you deploy your VaxVoIP-integrated SIP server on a computer equipped with a multi-core processor or CPU (such as core2duo, dual-core, quad-core, or hex-core), the VaxVoIP SDK intelligently distributes its processing load across all available CPU cores.
This innovative approach enhances operational efficiency and equips your server to efficiently handle a larger volume of SIP clients, ensuring seamless and high-performance VoIP services.
Multi processors support
The VaxVoIP SIP Server SDK offers robust support for multi-processor systems. When deploying your VaxVoIP SDK-integrated SIP server on a system with multiple processors, it leverages the collective power of all available processors, efficiently distributing its processing load across each core.
This intelligent allocation significantly enhances the performance and efficiency of your IP-Telephony network, resulting in a substantial gain. As a result, your CPU or processor is well-equipped to handle a high volume of SIP clients and concurrent calls, ensuring seamless and high-capacity VoIP services.
Play wave (.wav) files
Enhanced Call Experience with VaxVoIP SIP Server SDK: Our SIP server SDK introduces a set of functions for seamlessly playing wave (.wav) files during call conversations.
Leveraging buffered-based compression technology, VaxVoIP SIP SDK optimizes CPU resource utilization by compressing wave data only once and utilizing buffered data for subsequent playback. This innovative approach significantly reduces voice compression processing load on the CPU, thereby enhancing server efficiency.
Setting up call queues and playing music
Creating queues in various formats, including priority, circular, random, or custom, is a straightforward process. Additionally, you have the capability to play music during queue calls, enhancing the caller experience. For an in-depth understanding and implementation guidance, we invite you to explore our advanced SAMPLE CODES.
Establishing ring groups
Easily Implement Ring Group Functionality: Enable multiple phones to ring simultaneously when a single extension or number is dialed. Dive into our advanced SAMPLE CODES for in-depth insights and guidance.
Setting up pickup groups
Easily Add Call Pick-Up Functionality: Answer another person's call effortlessly. For in-depth guidance, download our advanced SAMPLE CODES and follow the instructions provided..
Call parking
Simplified Development of Call Parking: Create a convenient call parking feature, enabling users to place calls on hold at one extension and seamlessly continue conversations from any other telephone set or extension. For comprehensive insights and practical implementation, download our advanced SAMPLE CODES and follow the provided instructions.
Call barging
Effortless Development of Call Barging: Easily create call barging functionality, commonly used in call centers for training and supervision purposes. It enables call center managers to silently monitor live calls without the caller or agent being aware (known as silent call monitoring) and, when needed, join the call to communicate with both the agent and the caller (known as call barging). For comprehensive insights and practical implementation, explore our advanced SAMPLE CODES and DEMO.
Multi-user server side conference rooms
VaxVoIP SIP Server SDK: Enabling Server-Side Multi-User Call Conferencing and Chat Rooms. Our SDK supports the creation of server-side chat rooms by adding multiple calls to a single conference. For a deeper understanding and practical implementation, explore our SAMPLE CODE and DEMO for detailed insights and guidance.
Call hold
Easily Implement Call Hold with Music: You can effortlessly add a call hold feature to your SIP server developed using VaxVoIP SIP Server SDK. Customize the experience by playing music to the calls on hold, enhancing the caller's experience.
Call Transfer
Seamless Call Transfer Capabilities with VaxVoIP SIP SDK Integrated SIP Server: Our integrated SIP server offers the ability to initiate various types of call transfer operations, including blind, attended, and regular transfers. It effortlessly processes call transfer requests from diverse SIP clients, such as softphones, hardphones, ATAs, Wi-Fi phones, and more. For comprehensive insights and practical implementation, explore our SAMPLE CODES and DEMO.
Call Recording
Easily Integrate Call Recording in VaxVoIP SIP SDK Integrated SIP Server: Our SIP server seamlessly supports the addition of conversation recording features, allowing you to save conversations as wave (.wav) files. These capabilities are invaluable for the development of call center solutions, IVR systems, voicemail services, and more, facilitating the management and retention of call records.
DTMF tone generation
VaxVoIP SIP Server SDK: Exporting Functions for DTMF Digit Generation and Transmission. Our SDK supports three types of DTMF digit generation - RTP-based (RFC 2833), SIP-based (INFO), INBAND/Audio-Tone based. For a deeper understanding of these capabilities, Please refer to the TECHNICAL MANUAL for detailed insights.
DTMF tone detection
Real-time DTMF Detection with VaxVoIP SIP SDK: Our SDK is equipped to detect DTMF digits during call conversations and trigger corresponding events. It supports three types of DTMF detection - 1) INBAND/Audio-Tone based, 2) RTP-based (RFC 2833), and 3) SIP-based (INFO). These versatile features find application in the development of call center services, IVR systems, and a wide range of other applications. For comprehensive insights, explore our SAMPLE CODES and refer to the TECHNICAL MANUAL.
NAT and firewall friendly
Seamless NAT/Firewall Compatibility with VaxVoIP SIP Server SDK: Our SDK offers effortless connectivity for SIP clients, including softphones, hardphones, and ATAs, even when they are located behind NAT or firewalls. No additional settings such as STUN, E-STUN, or port forwarding are required, ensuring hassle-free connections to VaxVoIP SDK-based SIP servers.
SIP client authentication
Robust SIP Client Authentication with VaxVoIP SIP Server SDK: Our SDK offers comprehensive support for SIP client authentication procedures, enabling softphones, hardphones, ATAs, and Wi-Fi phones to securely authenticate and register with SIP servers developed using VaxVoIP SIP SDK. For in-depth insights, explore our SAMPLE CODES and refer to the TECHNICAL MANUAL.
Supported SIP based client and phones
Universal Compatibility with SIP Protocol: VaxVoIP SIP Server SDK seamlessly integrates with the SIP protocol, ensuring compatibility with a wide range of SIP-based clients, including softphones, hardphones, ATAs, Wi-Fi phones, and more. This compatibility enables clients to connect, register, and engage in two-way phone call communication with a VaxVoIP SDK-integrated SIP server.
SAMPLE SOURCE CODES
Select VaxVoIP SDK code samples that match your platform and language. These samples include SIP Server, IP PBX, IVR, and methods to Access the Audio data PCM (for developing AI-powered features and speech processing) of a VoIP call.
After downloading, unzip the files, review the ReadMe.txt for instructions, and refer to the TECHNICAL MANUAL for detailed guidance.
IP-PBX BASIC
Visual Basic .NET
Visual C#
Visual C++
Delphi
IP-PBX ADVANCE
Visual Basic .NET
Visual C#
Visual C++
Delphi
AUTO DIALER & IVR
Visual Basic .NET
Visual C#
We're delighted to offer a 30-day free trial of our product. You're welcome to download any of our sample code samples and give them a try. This trial period allows you to explore the functionality and capabilities of our product firsthand.
DEMO
Explore Our Demo Applications: You can access and download our demo applications, which have been developed using the VaxVoIP SIP Server COM component.
Take advantage of a 30-day trial period to download and utilize these demo applications, allowing you to experience the features and capabilities of our product firsthand.
IP-PBX BASIC DEMO
IP-PBX ADVANCE DEMO
Customize Your Solution with Sample Codes: If you find the need for additional features, you're welcome to download any of our sample codes and tailor them to meet your specific requirements.
Should you have any questions or require assistance, please don't hesitate to reach out to us.
DOCUMENTATION
Explore our Technical Manual: We offer a comprehensive technical manual that serves as a valuable resource for developers.
It provides an in-depth understanding of the internals and technical intricacies of VaxVoIP SIP Server SDK, enabling developers to quickly delve into the technical details.
TECHNICAL MANUAL (EXPORTED FUNCTIONALITY) Download Now
ACCESS AUDIO DATA (AI MODEL TRAINING & SPEECH PROCESSING) View Details
SIP CLIENT REGISTRATION PROCESS Download Now
SIP PHONE TO SIP PHONE CALL PROCEDURE Download Now
HOW TO CONNECT TO PSTN/GSM NETWORK Download Now
HOW TO CONNECT TO IP-TELEPHONY SERVICE PROVIDER (ITSP) Download Now
HOW TO ACTIVATE YOUR LICENSE KEY Download Now
DOWNLOAD
VaxVoIP SDK: Your SIP Server Development Solution. Our SIP Server SDK, inclusive of sample codes, a technical manual, the COM component (VaxTeleServerCOM.dll), and a demo application, provides a comprehensive set of tools for your development needs.
An All-In-One Solution for Developing a Robust SIP-Based VoIP Server: VaxVoIP SIP Server SDK offers a comprehensive package to create a fully functional SIP-based VoIP server.
Take advantage of our 30-day free trial by downloading VaxVoIP SIP Server SDK to experience its capabilities firsthand.
PRICING
We Provide Single Product/Software-Based Licenses for VaxVoIP SIP Server SDK: Our licensing model allows you to use one license with one of your products or software solutions.
With the purchase of a license, you will receive one year of FREE technical support, including access to new versions and upgrades. After the initial year of complimentary support, you can choose from the following support packages at any time.
VaxVoIP SIP Server SDK (SINGLE SOFTWARE LICENSE)
SINGLE SERVICE LIMITED CONCURRENT CALLS LICENSE
40 CONCURRENT CALLS Buy Now -$4,840 USD
80 CONCURRENT CALLS Buy Now -$6,880 USD
160 CONCURRENT CALLS Buy Now -$8,860 USD
320 CONCURRENT CALLS Buy Now -$12,680 USD
Technical support
SUPPORT PACKAGES
3 months support package Buy Now -$1,080 USD
6 months support package Buy Now -$1,660 USD
FAQs
How to get a trial?
Download the SDK, Demo Application, or any of our sample codes and feel free to explore them for a trial period of up to 30 days.
How to develop a SIP Server?
Simplified SIP Server Development with VaxVoIP SIP Server SDK: Creating a SIP server is made easy through the utilization of VaxVoIP SIP Server SDK.
Explore our website for accessible SAMPLE source codes, demo applications, and a comprehensive TECHNICAL MANUAL covering exported functions and events.
How can I access the PCM audio data of a call?
VaxVoIP offers two ways to access the audio data of a VoIP call in real time. For details, see the ACCESS AUDIO PCM page. Sample source codes also include implementations for accessing call audio data in real time.
Which audio codecs are supported?
We offer support for a variety of audio codecs, including GSM, iLBC, G711u-Law, G711a-Law, and G729.
Which operating systems are supported?
VaxVoIP SIP Server SDK is compatible with a range of operating systems. It seamlessly integrates with Microsoft Windows operating systems, ensuring compatibility with all MS Windows Servers and Desktop versions.
This versatile compatibility allows developers to leverage the SDK on various Windows platforms for SIP server development.
Full Support for Multi-Core Processors?
VaxVoIP SIP Server SDK is meticulously designed to harness the capabilities of multi-core processors, including core2duo, dual-core, quad-core, and hex-core systems. When running your VaxVoIP SDK-integrated SIP Server on a multi-core machine, the server intelligently distributes its processing load across all available CPU cores, resulting in significantly improved efficiency.
Full Support for Multi-Processor Machines?
VaxVoIP SIP Server SDK provides comprehensive support for multi-processor systems. When you deploy your VaxVoIP SDK-integrated SIP server on a computer equipped with multiple processors, the server effectively allocates its processing load across all available processors, resulting in a substantial increase in efficiency.
Is call recording supported?
Yes, our system allows you to record conversations, saving them in wave (.wav) file format.
Is sample source code available?
Yes, sample source codes in VC++, C#, and VB.NET are available on our website. Feel free to download any of these SAMPLE source codes and give them a try
Is demo application of SIP Server available?
Yes, please click the DEMO link.
Can I connect it with asterisk?
Yes, you can easily interconnect it with Asterisk and other third-party SIP-based servers, PSTN gateways, and gatekeepers. For detailed instructions and information, refer to our TECHNICAL MANUAL.
Can I develop PC to PC, and PC to Phone service?
Yes, developing PC-to-PC and PC-to-Phone services is straightforward. For a comprehensive guide, Please refer to the TECHNICAL MANUAL.
Can we use it in call centers?
Yes, it can be effectively utilized for the development of call center services, including call dialing, reception, queuing, and call transfers. For a deeper understanding, explore our SAMPLE source code and refer to the TECHNICAL MANUAL.
I want to queue incoming calls and play music to them, is it possible?
Yes, it's simple to develop the feature of queuing calls and playing music to them. For detailed insights, explore our SAMPLE source code for this functionality..
Do you sell the source code of SDK?
Yes, the source code of our SDK can be provided with user rights only. Additionally, training on the source code can also be provided.
Can I develop stealth listening feature?
For a comprehensive understanding of developing the stealth listening feature in your VaxVoIP SDK-integrated SIP Server, Please refer to the SAMPLE source code.
