VoIP SIP Client SDK 1.5

VoIP SIP Client SDK by conaito Technologies is a development toolkit that lets you embed reliable, standards-based voice calling into your applications. Built on the Session Initiation Protocol (SIP) and RTP media transport, it streamlines the creation of softphones, click-to-call features, and in-app telephony for business or consumer solutions.

The SDK includes an intuitive API, sample projects, and documentation to speed up integration. It handles core call flows—registration, call setup and teardown, hold and resume, transfer, and DTMF—while offering audio processing components such as echo cancellation, noise reduction, and an adaptive jitter buffer for clear, stable calls. Support for widely used audio codecs and call recording helps ensure interoperability and quality across networks.

Key capabilities:

• Standards-based SIP signaling with RTP media
• SIP registrar and proxy support
• One-to-one audio calling with call hold, resume, and transfer
• DTMF support for IVR and keypad interactions
• NAT traversal (e.g., STUN) to improve connectivity behind firewalls
• Audio quality enhancements: echo cancellation, noise suppression, jitter buffering
• Support for common audio codecs (such as G.711) and call recording
• Sample code, demos, and developer documentation for rapid onboarding
• Designed to integrate with popular programming environments

With its focus on interoperability, audio quality, and ease of integration, conaito’s VoIP SIP Client SDK helps you deliver dependable in-app calling while maintaining full control over your user experience and call flows. Features may vary by edition and platform.

VoIP SIP Client SDK is developed by conaito Technologies. The most popular versions of this product among our users are: 1.3 and 1.5.