How does Bandwidth support WebRTC?
WebRTC (Web Real-Time Communication) is a technology that allows browsers and other devices to interact and communicate with each other in real-time. It lets audio communication work inside web pages, eliminating the need to install plugins or download native apps.
Bandwidth's WebRTC API provides a fast and easy way to add voice to your application, backed by a nationwide direct-to-carrier network quality. If you’re already leveraging the power and flexibility of WebRTC or looking to start, you can now send your audio through Bandwidth!
What can Bandwidth’s WebRTC connection do?
Our audio-based WebRTC connection empowers users all over the globe to open up a browser (or an app) and get calling without requiring any telecom infrastructure. It also features:
- Reliable connection to PSTN
- 1-to-1 calling or conference calls
- Recording, transcriptions, text-to-speech, and more
- Variable bit rate, so you can stay connected even if your bandwidth decreases
How do calls get from the PSTN to WebRTC?
Calls come in from the PSTN to our Voice API and are sent to WebRTC via a Transfer verb. This is best accomplished via our SDKs because the call requires the inclusion of a JWT (a security token) as well as the Participant ID. The SDK handles getting the JWT and Participant ID into the right place.
If you’re interested in adding this feature, please reach out to your Account Manager. Not sure who your Account Manager is? Please open a ticket with your Bandwidth Support Team or hit us up at (855) 864-7776!
Ready to get started and looking for API instructions? Check out our WebRTC developer docs.