Telecommunications, much like the space program, has countless acronyms.  For Voice over IP systems, one of the most common abbreviations you will come across is SIP which stands for Session Initiation Protocol.  SIP is not only a protocol but the protocol for VoIP communications.  But what it is and what it actually does may sound quite vague and thus it’s often hard to understand its significance.

In this article, I hope to enlighten you about this mysterious protocol and how it fits into the big picture when it comes to modern voice communications.  Knowing about SIP is important for designing a network and even comes into play for making administrative decisions about the products and services you choose to purchase.  Even so, if for nothing else, I hope that you come to appreciate the innovation and elegance that this protocol brings to your business communications.

The anatomy of a VoIP communication

In order to understand SIP, we must first understand what VoIP is and examine the processes involved in making a VoIP telephone call.

VoIP Call Physical Components 

A VoIP telephone call involves various components including VoIP endpoints, which can be an IP phone that sits on your desk, an app in your smartphone, or even software on your computer or laptop.  Those endpoints are the actual telephone devices through which users make and receive calls.  These devices register to what is known as a call control server.  This server can be a company-owned device on your internal network or can be provided as part of a cloud-based VoIP service such as Ooma.

Parts of a VoIP Call

When a VoIP phone call is made, there are two distinct parts to the process.  The first is the transmission of the voice itself.  Voice is digitized, packetized, and transmitted over the IP network using packets, just like any other data.  Unlike a traditional telephone network, the voice doesn’t have to go through a central private branch exchange (PBX), but voice packets are exchanged directly between the two IP endpoints participating in the conversation.

The second part of VoIP communication has to do with call signaling.  This is the exchange of information that takes place between the participating VoIP endpoints as well as between the endpoints and the call control server.  Signaling is responsible for making phones ring, generating the various telephony tones we are familiar with, and implementing telephony features such as call hold, call transfer, call forward, and conference calling, to name only a few.

The SIP protocol is responsible for this call signaling of a VoIP call.  Somewhat counterintuitively, SIP does not carry any voice packets.  Voice packet transmission is the responsibility of other protocols such as Real-time Transport Protocol (RTP), which is beyond the scope of this article.

Voice and Signaling are two separate processes

Now it is important to note that the signaling and the voice transmissions are two distinct and separate communications over the same network.  The following diagram illustrates these communications:

Voice and Signaling

Note that the voice packets in red do not traverse the call control server but are exchanged directly between ext. 111 and ext. 222. However, SIP signaling is exchanged between the two IP telephones as well as between the phones and the call control server.

Additional scenarios

But SIP isn’t only used by IP phones and call control servers.  It can also be used by other devices that terminate VoIP calls including voice gateways.  Refer to the following diagram:

In this scenario, a call is being made from an internal IP phone to a destination on the Public Switched Telephone Network (PSTN).  A device known as a voice gateway will take part in the SIP signaling as well, in order to convert the VoIP telephone call to a signaling method used on the PSTN.

What does SIP do?

SIP, as its name suggests, initiates voice sessions.  But what does that mean practically?

SIP controls voice

It’s often difficult to get your head around the fact that that SIP itself does not carry voice packets.  Conversely, it is involved in the control mechanisms related to the initiation and termination of sessions needed to allow voice applications to function.  SIP, as it is defined by the Internet Engineering Task Force (IETF) in the RFC 2543 standard, defines the format of the control messages transmitted between participants in a VoIP exchange.  Call setup, maintenance and teardown, busy and ringback tones, and Dual Tone Multi-Frequency (DTMF) tones generated by the keypad and interpreted by devices, are all the responsibility of SIP.

Mimicking traditional telephony

These features are among those that have been employed in traditional telephony for decades and SIP essentially duplicates them within the VoIP domain.  Additional features commonplace on the PSTN and on conventional PBXs that SIP provides include call waiting, call hold, conferencing, call forwarding, call park, and a myriad of other telephony functions.  

SIP was designed to mimic the functionality of the PSTN and conventional PBXs to avoid the need of retraining users when moving from conventional to VoIP.  The idea was to allow someone to use a SIP-enabled telephone without any change in the tones, functionality, and general feel of the calling experience that users have become so familiar with over the years. 

SIP Registration

Beyond call control, SIP is also responsible for the registration of VoIP endpoints to the call control server.  It is for this reason that the call control server is often referred to as the SIP server.  SIP has a client-server architecture and primarily functions with a registration mechanism where SIP clients, such as an IP telephone or software running on a PC or a mobile phone, register to a SIP server.  

Once registered, a SIP client will be able to make calls based on the allowances provided by the configuration of the SIP server.  Depending on how the SIP server is configured, calls can be made either using a string of digits, just like traditional telephony, or by the SIP Uniform Resource Identifier (URI) which is used as the username of the SIP client.  The SIP URI has the form of sip:username@host where the host is the IP address or DNS name of the client.

Introducing advanced features

However, SIP is not limited to just reproducing features available on conventional telephony systems but was designed to go beyond that and to incorporate advanced features and functionalities that take advantage of the IP infrastructure upon which SIP is based.  This is why organizations have been adopting SIP more and more as the voice protocol of choice within their private networks.  

VoIP systems based on SIP can easily expand VoIP network services by adding features such as video, presence, and remote users to their existing infrastructure with very little intervention into the existing system.  For example, complex contact center features and functionality like those provided by Freshcaller, can be created in minutes.  API integrations, like those offered by Aircall, allow the interoperation of the phone system with other productivity software like CRMs, e-commerce, and helpdesk platforms, which can streamline business processes and improve employee efficiency.  

All of this is made possible because SIP is the standard control protocol that is used by the vast majority of modern VoIP software, hardware, and services.


The act of picking up the phone and dialing has been performed by humans for over sixty years.  It’s rare that anyone thinks about what’s going on in the background to successfully complete a call.  But SIP takes care of all of that.

SIP is truly an exceptionally well designed, flexible, and scalable protocol for IP voice and media in general.  It delivers a user experience almost identical to the PSTN while at the same time, it offers a myriad of additional features and services that are useful, beneficial, and in today’s often chaotic business world, invaluable.  

Because of this, there is no discernable end to the use of SIP as the de facto VoIP standard, nor has there emerged any other protocol that has threatened its rule in the least.  While traditional telephony protocols and services are slowly but steadily on the decline, SIP’s future seems to be secured for at least a generation.  And in an industry where changes occur at a tremendous speed, SIP’s apparent longevity can be considered almost an eternity.