Connecting WebRTC endpoints and Telephony Systems with OpenTok SIP Interconnect

OpenTok SIP Interconnect Webrtc for telephony systems

As we continue to work towards enabling developers to reap the full potential of WebRTC, we wanted to demonstrate connecting a WebRTC audio stream with a PSTN user, using OpenTok SIP Interconnect and a third party SIP-PSTN Gateway.

OpenTok SIP Interconnect enables interoperability between WebRTC endpoints and existing telephony systems. This allows users to make SIP-based audio calls, while simultaneously browsing the website or mobile application. The OpenTok SIP Interconnect Gateway is composed of two parts: the SIP Interface and the OpenTok Interface. Below you’ll find a graphic detailing how communication works between a WebRTC app and SIP endpoints:

OpenTok SIP interconnect WebRTC to SIP

There are many benefits of using OpenTok SIP Interconnect, such as integrating with voice powered by WebRTC to offer real-time customer service or using it as a PSTN fallback in case there are connectivity issues or incompatibility between browsers. Additionally, end-users who are offsite can easily communicate by dialing into OpenTok sessions through a SIP-PSTN Gateway. To find out about more use cases, read our Introducing SIP Interconnect blog.

For supporting material, we’ve created an OpenTok-SIP-Samples repo that showcases the following samples:

Please keep in mind that the SIP Interconnect API does not support direct incoming SIP calls. However, you can implement a conferencing solution where a user dialing in from a regular phone (PSTN) and a user dialing-out using SIP Interconnect from an OpenTok Session is bridged by a SIP-PSTN gateway.

If you’re interested in learning more about how OpenTok SIP Interconnect works, please reach out to me. I’d love to know what you are working on!