The era of clunky, keypad-driven legacy IVR customer service systems that have long frustrated users is finally over. The future of Interactive Voice Response is truly conversational, and it’s ready for prime time. That’s why Deepgram’s State of Voice AI 2025 report says 84% of business leaders
Whether you’re building a VoIP application, a video conferencing platform, or any real-time communication solution, ensuring optimal performance in environments with slow or unreliable networks can be a significant challenge. This is especially true for mobile networks and satellite communications, where latency, packet loss, and bandwidth limitations
Session Initiation Protocol (SIP) and WebRTC are both essential technologies in the field of real-time communications, particularly for voice and video over IP. While they serve complementary roles, they operate differently and have distinct functionalities. In this post, we explore how to architect the integration of WebRTC
The seamless browser to browser real-time audio and video communication that WebRTC enables is supported by a complex infrastructure. Things like signaling, NAT traversal and codec optimization can be difficult to maintain by the average development team unfamiliar with the intricacies of the WebRTC stack. CPaaS providers
