SIP, IP and VoIP
What are the differences between SIP and IP telephony? In fact, comparing them is not entirely correct. It’s like comparing a “laptop” and a “computer”: that is, the categories are somewhat different. Let’s take a closer look.
But first, it should be noted that when people talk about IP and SIP telephony, they often mean the same thing. It just so happened that these terms are identical and are used interchangeably. So don’t be surprised if you realize that your interlocutor is actually talking about SIP, but uses the term “IP telephony” in speech.
IP telephony: it’s something broader than SIP
IP telephony is a type of telephone communication. Its main feature is the method of data transmission. This method implies the transmission of data over IP networks, and not via traditional telephone lines. IP is also a set of rules that regulate data transmission over a network.
The principle of IP telephony can be understood by looking at the stages a signal goes through during a communication session:
- Conversion of the analog signal into digital. This is handled by a digital-to-analog converter.
- Encoding the digital signal for transmission over the network. This allows devices/programs on both the caller and recipient sides to “speak the same language.”
- Dividing the data into packets. This is necessary to simplify and optimize the transmission of information over the IP network. If data were transmitted as a whole, any error would require retransmitting everything. If the information is divided into packets, each contains a portion of the data and has a unique identifier. If a packet is lost, only that specific one is resent.
- Transmitting packets over the Internet (or other IP network). Packets are delivered to the recipient through the network.
- Extracting data from packets and decoding the information. The received packets are unpacked and transformed back into voice.
- Playing the voice on the recipient’s device.
So, IP is a type of communication. And for its implementation, protocols are used. And SIP is one of those protocols.
SIP is a “part” of IP telephony
SIP is one of the protocols upon which IP telephony operates (used to deploy the network and establish communication). It is a session initiation protocol. Currently, it is one of the most popular and widely used in IP telephony equipment.
SIP has certain similarities with HTTP, since it is also text-based and uses a constant back-and-forth of requests and responses. This contributes to relatively simple deployment and configuration of telephony.
Some features of the SIP protocol for IP telephony:
- Transfer of headers in ASCII format, which allows for relatively easy reading on different clients.
- Compatibility with many data transmission protocols. SIP itself does not transmit voice, video, or other media; this is handled by other transport protocols. SIP works seamlessly with many of them.
- SIP does not require centralized control, allowing the creation of flexible and scalable networks.
- This protocol can be used to set up IP telephony in various network types, including corporate and private networks (thanks to NAT support).
- Relatively high level of security. SIP supports data encryption and authentication mechanisms.
- DNS support for finding session participant addresses, simplifying network setup.
SIP and IP telephony: so, is there no difference?
Turns out that comparing SIP and IP is incorrect. One could say that SIP is a “component” of IP. And comparing the whole with one of its parts is not exactly right or appropriate. Typically, when we talk about IP telephony, we’re also referring to SIP telephony.
On the other hand, if in a particular context IP telephony implies the use of another protocol instead of SIP, then we can talk about some comparison.
So what can SIP be compared with?
Protocols used in IP telephony are generally divided into two groups: signaling and data transmission. SIP belongs to the first category. That means it should be compared with other signaling protocols.
Signaling protocols for IP telephony perform the following functions:
- Establishing and terminating calls
- Registering an IP phone, software or other client on the provider’s server
- Call forwarding
- Exchanging information about numbers, subscriber names, etc.
There are several signaling protocols used in IP telephony.
SIP and IP telephony: but there’s also a third “player”
Since we’re talking about the differences between SIP and IP telephony, it’s also worth mentioning VoIP (to get completely confused — or, better, to finally understand it all).
So, IP telephony is a general term for phone communication organized through IP networks (Internet, corporate, private, etc.).
Another key characteristic of IP telephony is the type of transmitted data — it refers only to calls (i.e., voice transmission) and video communication.
VoIP, however, is a broader concept that refers to the transmission of any voice-related data.
That is, VoIP enables phone calls over IP networks, video calls, webinars, video conferences, online streaming from IP surveillance cameras with sound, and more.
SIP and IP telephony
It’s clear that SIP and IP telephony have become synonymous. Many people use them in the same context.
At the same time, as we’ve seen above, they are somewhat different.
Still, these terms are often used to mean the same thing.
Context matters when using these terms. And it’s important to understand that there is a difference:
SIP is a protocol, and IP telephony is a method of organizing communication via IP networks.
Besides SIP, IP telephony also uses other protocols.
Some are alternatives to SIP (handling similar functions, like H.323), and some work together with SIP to handle different tasks, such as managing or transporting the communication session.
In addition to the protocols already mentioned, you may encounter others when working with IP telephony. Among them are:
- MGCP (used to control a media gateway)
- SCCP (developed by Cisco to manage its devices)
- RTP (provides voice and video transmission in VoIP networks in real time)
RTCP (works alongside RTP and ensures the exchange of data about transmission quality during VoIP communication)
