Product

Siprix VOIP SDK is a reusable component that helps developers build applications with voice and video calls over the internet. It provides It provides ready to use building blocks, allowing programmers to integrate calling features seamlessly into their apps without writing complex code from scratch.
The VOIP SDK handles call initiation, connection management, and disconnection, freeing you from manually implementing these tasks.
The result is faster development times, reduced costs, and the ability to focus on unique features and functionalities within your application.

SDK is built from the same C++ code base, providing a consistent API and behavior across all platforms. The key benefit of this approach is cost-effectiveness. Creating a single application that runs on multiple platforms, companies can save significant costs by reducing the time and resources required for development, enabling faster deployment and time to market and simplifying maintenance.

SDK allows multiple SIP accounts with independent configurations to be added and used simultaneously. Each account can have its own transport, port, media and encryption settings. This approach allows organizations to segment their communications according to specific rules. For example, different SIP providers offer different rates, especially for international calls. By using a mobile application that can manage multiple SIP accounts, you can choose the cheapest provider for each call, potentially resulting in significant savings.

SDK offers advanced call management capabilities. It can establish and manage concurrent calls, play audio from files, seamlessly switch between active calls, and merge them into a conference. This functionality empowers developers to implement features like call waiting, three-way calling, and music on hold, enhancing the user experience for communication applications.

The SIP protocol provides different ways of communication between endpoints and also the ability to negotiate supported features and the most appropriate format. As a SIP client, the SDK implements many of them, including:

Audio Calls

Video Calls

Text Messages

For modern iOS applications offering incoming (VoIP) call functionality using CallKit and PushKit frameworks is practically mandatory. The SDK eliminates the need to find and integrate third-party components and provides a ready-to-use implementation that can be easily customized or extended and included in the final application with minimal effort.

In today's VoIP applications, securing both media and signaling traffic is no longer a luxury but a necessity. Our SDK provides a comprehensive implementation that includes:

Signaling Encryption (secure call initiation, routing, and termination using TLS)

PBX authentication (verify PBX's certificate before connecting it)

Media Encryption (secure media streams using industry-standard SRTP).

Dual-Tone Multi-Frequency (DTMF) allows users to transmit digits using the familiar keypad interface, facilitating navigation through automated systems and IVR services (pre-recorded menus like “press ‘1’ for ‘...’”). SDK supports sending/receiving DTMF in two different ways as RTP signalling packets and also as SIP INFO requests.

These features empower users to include additional participants in ongoing conversations. Calls can be forwarded to another number or seamlessly transferred to a colleague, allowing quick collaboration and efficient problem-solving. Our SDK has the ability to make blind transfer to any phone number, attended transfer between two connected calls and also redirect incoming calls without answering them.

VOIP applications typically register with a PBX to receive incoming calls. However, network conditions are not always stable, and users may move between different networks (e.g., switching from Wi-Fi to cellular data) or experience temporary disruptions in connectivity. SDK is able to detect network changes, notify the application about them and also recover registration when connection is restored. All these steps work automatically and don't require additional developer efforts.

Our VoIP SIP SDK offers high-performance echo cancellation, enabling seamless voice transmission across devices. By using advanced algorithms, it detects and eliminates the echo, ensuring clear and uninterrupted communication. Echo cancelation is especially important in environments with delays, improving the overall call quality and user experience.

The audio mixer plays an important role in effectively managing and processing multiple audio inputs. SDK has its own implementation that allows to switch between multiple connected calls, join them into a conference and also play sound from files along with sound from microphone.

The model serves as a data abstraction layer, encapsulating the information that drives the UI. Modern frameworks use the data binding mechanism that triggers re-rendering the UI whenever a property within the model is modified. This eliminates the need for manual UI manipulation based on data changes, promoting cleaner and more maintainable code. SDK doesn't have its own UI and provides ready-to-use models. One side of the model interacts with the API (controls call/account states) and another side provides data (state) used by the application for rendering.