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
In this post we are going to use the Janus SIP gateway plugin to build a WebRTC to SIP / SIP to WebRTC communication and monitor it with Homer. If you need media server capabilities don’t build things from scratch. You can use one of the most
