Some time ago we explained in our blog what are SIP calls and how this technology works. However, we did not go into the in-depth analysis of the base protocol that providers of this service usually use, that is, the SIP protocol. Below we will see what it consists of and what do you need for its functioning, in addition to the possible alternatives you have if you are thinking of switching to VoIP telephony.
What does SIP protocol mean?
The SIP (Session Initiation Protocol) is defined as a set of signaling rules that is responsible for establishing the communication between a user and the server, so that a link is established between two (or more) devices.
In a technical way, this protocol is used to establish, modify and even end communication sessions that are carried out through an IP network and the SDP protocol or Session Description Protocol in English. However, it is important to say that SIP relies on other protocols to achieve that connection. In fact, it does not transmit or receive any audiovisual media, but only initiates or terminates a communication.
On the other hand, if we simplify the process and the names used, we can see that it is something as simple as the communication between two people through a telephone, since there is a connection between these two points. Or we could even think of more complex formulas such as a videoconference between several participants through the web, and in this case there would be a union between several points.
IP telephony is the area in which this protocol is most interesting and used, because as it is an open standard, it is very attractive in the telecommunications market, so that in the last decade sales in this sector have increased exponentially.
Its main functions, although we have already mentioned some, are:
- Locate and register participants.
- Manage the components of the system, as well as the group of participants.
- Description of the characteristics of the sessions that can be carried out.
What do I need for SIP communication working?
To adequately explain SIP protocol communication, we will focus mainly on making and receiving telephone calls over the Internet, which is known as Voice over Internet Protocol or VoIP. The reason is that the Internet is made up of a network of different protocols, but the SIP protocol is the most used in making this type of calls, together with WebRTC, which we will discuss later.
In order to make such calls, you need to have:
- A SIP address that is contracted through telephone service providers.
- Installation of a software or softphone. This is key, since if you do not download the application or program on the device from which the calls will be made, you will not be able to establish communication.
- An internet connection, preferably by cable, because to have a proper communication is necessary a high quality bandwidth and a stable connection, especially for video calls.
- Share the SIP address. The other person needs to know the address, just as the phone number or email is shared.
Once we have made the necessary installations, we can start using this technology. To this end, providers of these services connect one or more channels to the customer PBX. The telephone numbers are then linked on a telephone line (SIP trunk). When a call is made, SIP establishes the connection with the receiver’s device, to make way for other protocols that carry the content of the call (audio, video, etc). When the call ends, the SIP protocol is responsible for this task.
Is there an alternative to SIP protocol?
The answer is yes and it’s called WebRTC. We cannot deny that, at the time, the development of SIP protocol was a revolution in the world of telecommunications and although it has many advantages, WebRTC technology goes a step further.
Both are used to communicate, so that any company can use them to improve their customer service and be able to connect to any device if there is an internet connection, allowing them to save effort and time. However, the most important difference between both is that those companies that operate through SIP require, as mentioned above, the installation of a software, so it means downloading it every time you want to change your device and need constant updates. With WebRTC, this problem disappears, as it is only necessary to access the browser from any device.
Because it operates over the Internet, both protocols are cheaper than traditional telephony. But it is true that SIP has higher costs than WebRTC. The reason is that while the first one a software is required, a special equipment that allows its operation (as it only allows you to make calls from a special softphone or IP phone) and investing in maintenance costs, with the first is not necessary, due to the fact that the only requirement is to have a computer, tablet or mobile to start working. And who does not have any of these instruments today?.
Related to the latter, SIP allows less mobility and flexibility, since it requires the special teams we talked about, it is essential that the team moves with you wherever you go, something that in many cases is impossible. For example, these days Covid-19 crisis has forced many companies to opt for telecommuting, which means that, if you use SIP protocol, you are forced to have the necessary equipment installed at home or use softphones that do not guarantee the required quality all the time. Instead, with WebRTC, you simply need your device with an Internet connection, thus providing easy, complete and HD quality use in your calls.
At Fonvirtual we offer you our call service through WebRTC technology. If you are thinking about switching to VoIP, this is the best option, since our service also offers very useful tools that will help you to optimize your customer service and to take a step further in your enterprise communications. If you still have doubts about this technology, do not hesitate to contact us. Our team will advise and guide you to choose the model that best suits your needs.