So you’ve been tasked with learning SIP to support and extend your company’s VoIP telephony and Unified Communications (UC). But where do you start? An understanding of basic SIP terms is key.
What is SIP, and why does it matter?
What is SIP? The clue is in the name: Session Initiation Protocol. SIP is a peer-to-peer protocol standard developed by the Internet Engineering Task Force (IETF), used for initiating—and also managing and terminating—voice call sessions over the Internet.
It’s broader than that now, though. SIP can be used for initiating, managing, and terminating other real-time communications, like video and text, as well as voice, and can be used over any IP network, not just the Internet. It’s also extensible: SIP can manage other applications, too.
SIP matters because many organizations have relatively high-bandwidth WAN connections, and an increasing number use SIP to carry voice as well as data traffic over those connections.
Typically, organizations do so using a Voice over IP (VoIP) switch or SIP gateway on their premises, using their WAN connection to run a SIP trunking service, which carries VoIP calls and ultimately connects them to the public switched telephone network (PSTN).
That’s not the only way companies can benefit from Internet-based telephony, though. They can also use SIP to set up calls from one IP telephony device directly to another—directly in the sense of cutting out the PSTN altogether.
This leads to confusion about what, exactly, SIP is. People often use the terms SIP and VoIP interchangeably. So, just to clarify, VoIP is the practice of using IP networks to make voice calls by breaking up the conversation into packets, which are transmitted between the calling parties. SIP is the protocol that initiates those calls.
How SIP works
When someone picks up an IP phone and tries to make a VoIP call to another person using their SIP address, it is SIP’s job to find out where in the world that person is and to what network they’re connected. SIP can then send a message to the IP phone with the caller’s SIP address, which makes it ring. When the call recipient picks up the phone, SIP then does all the network housekeeping required to set up the call, which involves ensuring both parties are using the same codecs to turn voice into IP packets and back to voice, and also ends the call when either or both parties hang ou.
It sounds simple, but there’s a lot going on behind the scenes. So let’s take a closer look at the various parts of a system that are involved in using SIP to make a VoIP call.
Location Service: The location service gets information from a registrar server about where a user is and provides that information to a SIP proxy server or redirect server. That way, a proxy server that receives a request to start a call with a user knows where to direct the request so that it reaches the user.
Redirect Server: When a SIP user is not on their own network, a redirect server sends SIP requests to the network to which they may be connected. Effectively, it maps a SIP address to an alternative SIP address and returns that address to the caller, who can then attempt to reach them at the alternative address.
Registrar Server: When a user switches on their device and connects to a network, effectively coming online and making themselves available to make and receive calls, there needs to be a mechanism for transmitting that information to other people. The user does that by authenticating and registering with their registrar server. In technical terms, they do that by sending a register request to the server. The server then stores the information it receives, such as the user’s IP address, so that others can find them. Typically, the server will also forward this information to be saved by location services and redirect servers.
SIP Address: A SIP address identifies and locates each user of a SIP-based VoIP system. The addresses look like email addresses in that they have two parts, separated by an @ symbol—e.g., [email protected]
SIP Proxy Server: When a user agent attempts to connect with another user agent, it sends out a request, which usually goes to a local SIP proxy server. The proxy server acts as a proxy for the user agent, handling the request and sending it on to the proxy server belonging to the call recipient’s organization. Essentially, the proxy server acts as a call router, forwarding on a call request to another proxy server closer to the recipient. A SIP proxy server can also be used to enforce corporate SIP policy. For example, it may forbid certain callers from using SIP to make calls outside the organization.
STUN server: STUN stands for Session Traversal Utilities for Network Address Translation (NAT). A STUN server is designed to allow a SIP user to discover the public IP address and port number that their VoIP application uses. The purpose of this is to overcome the problem of effectively being cut off from traffic due to being connected to a network behind a NAT router.
User Agent (UA): This is the SIP software that sits on endpoints like IP phones, or runs in IP telephony software on a computer or mobile device, sitting between the end user and the network. A user agent can act as a client or a server, switching between the two roles as necessary.
User Agent Client: A user agent acts as a user agent client when it generates requests, such as when it starts the process of setting up a call to another person.
User Agent Server: A user agent acts as a user agent server when it receives a request and processes it, such as when another person is trying to call. If user agents are on a small local network and know each others’ Uniform Resource Identifier (URI), then they can send and receive requests between them, and a call can be established without any further SIP infrastructure. This is rarely the case, however. That’s why SIP has a mechanism for finding the call recipient, regardless of where they are and to what network they happen to be connected. But doing so involves an infrastructure of SIP servers, such as the SIP proxy server, registrar server, redirect server, and STUN server discussed above.
So there you have it: some of the key terms you’ll need to understand SIP. Stay tuned for future installments in our Unified Communications 101 series, designed to help you make the most of your enterprise’s communications infrastructure.