SIP Basics

What is SIP

Sip is a peer to Peer App layer(7) protocol used for signaling and setting up sessions. It's independent of the underlying transport so it can be TCP,UDP, MPLS, ATM and so on. Sip is also text based so its easier for us to read. Sip runs usually on port 5060 for connection to SIP severs and endpoints. We use SIP to setup and Tear Down Voice, Video, or any application that requires a session. The actually communication is done over a different transport which is usually RTP in the case for VOIP.

Why do we use SIP?

SIP is here for us to provide PSTN like features in a IP based world. Sip provides the call setup and signaling for this to happen. So when you pick up your phone to call someone, when you dial a number, when you hear ring, back and busy signals.


SIP components

SIP has 5 parts that play a role in getting it up and running

USER Agent

VOIP phone, Cell Phone, Video Terminal. These devices can setup and Terminate sessions when they exchange the correct message. Subsets of these include User agent Clients which can send SIP request, and User agent Servers which will act as a "proxy" and acts on the behalf of the user


Proxy Server

This acts as a server and client for making request. The request can be send on to external devices or acted upon internally.

Redirect Server

They take SIP request, Give it a address and returns them to the client.

Registrar.

Server takes request for updated the database.



B2BUA

Back to Back User agent.

Operate both endpoints of a phone call, Acts as a user agent server and client and generates request and responds.



Other things to know:

Sip functions just like HTTP in how it was made. As I said above its Human Readable text.



SIP Messages


2 types that we have

Request and Responses

Request are sent from the client to the server, and responses are sent in the opposite direction.

Sip request messages

"INVITE" which tells you someone is trying to setup a call with you.
"ACK" A final Response to a "INVITE"
"BYE" Ends a call
"CANCEL" Just like it says cancels a search
"OPTIONS" Queries capabilities
"REGISTER" Registers the User Agent with the Registrar

SIP also Has response messages which are like the usual HTTP errors we see like the 404 that we get when browsing


SIP address

Fully Qualified Domain names, E164 Address or mixed address

So how would we configure such a think on our gateway and Call Manager?

Stayed Tuned

There is a lot of flexibility so in the future we are going to cover setting up SIP to a ITSP, Setting up SIP for functionality in CUPS, CUE and possibly some SIP 3rd party phones.