Build an integrated experience within-browser calling
Bandwidth’s WebRTC Calling API (Web Real-Time Communication) enables you to quickly add voice calling to your web-based applications without the need for any telecom infrastructure.
Enable the in-browser calling functionality that your app needs, without the hassle of deploying expensive telecom infrastructure.
Our server-side SDK supports all of Bandwidth’s standard languages NodeJS, Java, Ruby, Python, PHP, and C#.
Yes, we have a Hello World application that shows how to connect someone in a web browser with someone calling from the PSTN.
All PSTN interconnection is facilitated by our Voice API offering. Our WebRTC Calling API uses a combination of SDKs and Voice API functionality to integrate calling to the PSTN.
As with many offerings, our WebRTC uses a proprietary protocol over websockets for browser signaling.
Yes, you can pin up an agent for many hours and bring callers into the session, removing them when they are done and then bring in subsequent callers.
However, we have created a solution that makes it easy to create new sessions, so we do recommend this, as it makes for easier auditing and cleaner billing after the fact.
For both inbound and outbound calls, you just transfer the call to a special SIP URI. There is a helper function that does this for you in the SDK.
Yes! You can do this using our Voice APIs, then transfer the call to WebRTC when you are ready.
Nothing. The Bandwidth client SDKs, native browser capabilities, and use of network media servers remove the need for our customers to manage STUN, TURN, and ICE.
Yes, WebRTC calling is compatible on mobile and desktop browsers.