SIP stands for Session Initiation Protocol and it’s the protocol that allows VoIP calls to be made and received over a data connection. SIP communicates using ports 5060 & 5061 on the Internet; you can read more about ports here
The exact specification for the core of SIP can be found on the Internet Engineering Taskforce’s website and the latest thinking and workings on it can be found on the Working Groups page.
SIP can’t make a VoIP call work on its own. SIP just takes care of introducing and closing the phone call so to speak. It does that through two stages:
1) A SIP registration to the Yay.com Platform so we know you are connected to us and ready to make and receive calls
2) A SIP Invite conversation when you actually make and/or receive a call
We’ve posted some diagrams of the exact conversations that take place below to help you understand the steps involved. These steps in each case take place in milliseconds. A brief outline of what they do is below.
The SIP registration
This takes place when your softphone or hardware phone registers with us. If you like, its you saying “Hi, we are here and ready to make and receive phone calls”.
Most softphones and hardware phones register with a session time. That can be whatever length you choose it to be and is commonly set between 1 & 60 minutes in the preferences/configuration of your softphone or hardware phone.
When that period finishes, it simply registers again with us. It’s VoIP's way of saying, "hey I’m still here, just wanted to check in with you again so you don’t just presume I’m still here!". There is no loss of continuity on your ability to make or receive calls while its doing that routine “check in”.
Below is a diagram that shows the exact steps of that.
The SIP Conversation
This takes place when you actually make or receive a call and the “Invite” part is the most important part of it.
The “Invite” is where your client and the Yay.com servers are negotiating a little. You are telling us what Codec’s your softphone or hardware phone supports and in what order of preference and we are doing the same. We are also agreeing what RTP ports we are both going to use to transmit audio to/from each other.
Below is a diagram that shows the exact steps of that: