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:
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!
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