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.


  • 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


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

complete setup

Once everything is filled out click the OK button and set your name. Click the -> arrow to load the WebRTC Demo

Demo GIF


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