JSSIP with Bandwidth Voice API
⚠️ Bandwidth no longer supports WebRTC per rtcpMuxPolicy
Older versions of chrome may still work. Please check back later for more information or contact sales to check out status.
Prerequisites
- Register for a Bandwidth Voice API account here
- Follow the SIP Guide to create a server to handle SIP incoming calls and PSTN Calls
- If you want to make calls to the PSTN (normal phones) you will need a server to handle events from Bandwidth
- Make phone calls
For a more in depth guide, view this article
Quick Start with JsSip
Once you have stood up a server to handle callbacks from Bandwidth AND created a domain with at least one endpoint that is associated with an application; you're ready to get started with SIP
https://tryit.jssip.net/
| Name | Value |
|---|---|
| SIP URI | sip:{endpoint_name}@{domain_name}.bwapp.bwsip.io |
| SIP password | value used when creating the endpoint |
| WebSocket URI | wss://webrtc.registration.bandwidth.com:10443 |
Once everything is filled out click the OK button and set your name. Click the -> arrow to load the WebRTC Demo
Demo GIF
