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
For our 70th episode of WebRTC Live, Arin welcomed Kamailio Consultant, VoIP Engineer, SIP Expert Fred Posner to discuss bridging WebRTC to SIP via Kamailio and use cases such as call centers, remote workers, and PSTN connectivity.
There are a variety of ways you can use traditional telephony in your WebRTC video app, as well as different architectures you can choose to support it from commercial to open source. Let’s explore the reasons you might want to integrate a dial-in or dial-out capability into your WebRTC video or audio application and look at a sample architecture.
WebRTC & SIP: The Demo! WebRTC and SIP are two of the most important technologies in today’s real-time communication ecosystem. Session Initiation Protocol (SIP) is heavily used in VoIP technology; webRTC is used for browsers, mobile devices and native communication capabilities without additional software plugins. In this
