I’m working on a feature for my website that allows users to make calls to public network numbers (e.g., customers) using WebRTC. I’ve been provided with a username (which is a number), a password, and a domain by Service Provider.

Here are the details:

Credentials from the Service Provider:
Username, Password, Domain

Requirements:
Use WebRTC to handle the media streams (audio).
Integrate with a SIP server to manage call signaling.
Make and receive calls to/from public network numbers.

Specific Questions:

SIP Server Configuration:

How should I configure Asterisk or FreeSWITCH to work with the credentials provided by my Service Provider?
Any specific settings or modules required to facilitate WebRTC calls?
JsSIP Integration:

How do I properly configure JsSIP with the service provider credentials to make outgoing calls?
Example configuration and call initiation code snippets would be highly appreciated.
NAT Traversal:

What are the recommended STUN/TURN server configurations for ensuring reliable media path establishment?
General Best Practices:

Any best practices or potential pitfalls to be aware of when integrating WebRTC with a SIP server for this use case?

I have basic knowledge of WebRTC and SIP, but I am not very experienced with network protocols.
I am looking for a comprehensive guide or examples to ensure a smooth integration process.
Any guidance, examples, or resources would be immensely helpful. Thank you in advance!