There’s one important catch to telehealth: not everyone has reliable internet access, and connectivity can fail at any point during remote medical appointments, even in regions with strong coverage. Telephony integration using traditional voice networks remains a vital backup solution when digital connections drop during virtual healthcare
Session Initiation Protocol (SIP) and WebRTC are both essential technologies in the field of real-time communications, particularly for voice and video over IP (VoIP). While they serve complementary roles, they operate differently and have distinct functionalities. In this post, we explore how to architect the integration of
Real-time communication has become an essential part of the way we interact with each other over the internet. From video calls and live streaming to interactive gaming and instant messaging, we are constantly relying on reliable and instantaneous exchanges of information. Enter WebRTC: a transformative technology standard
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
