Reputation: 86
I am working on the webRTC application. Which can receive a call from browsers, The caller's source can be from any phone number or the extension dialed from the webRTC application. I am using the FreeSwitch server for this purpose.
Can anyone help me to know if this is achievable using only WebRTC or do I need SIP + webRTC like sip.JS, jsSIP
Upvotes: 0
Views: 294
Reputation: 30714
You can create a calling application using WebRTC without SIP but you will need to create or choose some form of signalling protocol. WebRTC can transport the audio and video packets for you but it does not specify how to set up the connection between two peers.
Given you're intending to use FreeSWITCH you may find that using SIP is the easiest option for you. FreeSWITCH plus one of the SIP javascript libraries you've mentioned solves your signalling requirements.
Upvotes: 2