I am new to the topic, so pardon my lack of understanding.
My company has an established SIP stack and we want to add Video Conferencing capability to it.
As per my current understanding, there are at least two ways how Video Calls/Conference can be established:
- Since SIP connection is format agnostic, it is possible to use SIP both for signaling and audio/video transport via RTP
- Another way is to use WebRTC in conjunction with a gateway to route media stream via SIP (does it make sense?)
Assuming I understand the options correctly, the questions are:
- Can I make a video call from WebRTC based solution to a SIP-only client via Asterisk?
- Do I need a WebRTC gateway to connect to the Asterisk server in its default configuration?
- What is the purpose of PjSIP and alike?
Thanks in advance for the clarification!