Session initiation protocol (SIP) is the protocol used in VoIP communication to make voice and video calls, mostly for free. SIP is widely used to control Internet multimedia conferencing, Internet phone calls, and multimedia distribution in both the core and periphery of the communication network.
If you want to take a closer look at the question “what is SIP calling?” then this article is for you.
What Is Sip Calling
SIP, also known as Session Initiation Protocol cluster. Thus, SIP call is not a conversational protocol but rather a digital system that allows devices to interact with each other. It is similar to a computer’s Hardware, setting its standards for connecting devices.
Understandably, SIP is one of the protocols used to implement VoIP, namely establishing and terminating audio and video calling sessions. Currently, SIP is the most widely used protocol and the most standard for businesses. Therefore, its coverage level is extremely large.
SIP calls are primarily text-based, and request-response mechanisms make troubleshooting easier. The actual data transmission is performed by the Transmission Control Protocol (TCP) or the User Data Packet Protocol (UDP) on Layer 5 of the OSI model. The Session Description Protocol (or SDP) controls which protocol to use.
The SIP describes the call participants’ identity and how the participants can communicate over an IP network. Packaged inside the SIP messages, sometimes, we may also see an SDP declaration.
But what is an SDP? The SDP (Session Description Protocol) determines what type of communication channel will be set up for the session. This will usually describe what kind of codecs are available and how communication tools can reach each other over an IP network.
When this call exchange is completed, the media is transmitted using another protocol, typically known as RTP or Real-Time Transfer Protocol).
Read more: What Is SIP ALG?
What is A SIP Phone?
SIP Phones are like VoIP Phones or softphones. It uses the Open SIP system to operate and control phone calls. IP-based network using another RTPR or Open Standard will incharge of transmitting the actual voice. Because these protocols are collectively referred to as VoIP (voice over internet protocol), these are devided as VoIP Phones or VoIP Clients.
There are two types of SIP phones:
This is the same phone as a landline phone but can receive and make calls over the Internet instead of a traditional PSTN system.
Hardphones use network-aware components or, more specifically, IP-aware components. They will connect to the IP Network using a regular ethernet cable or using WiFi.
This type of phone runs on software. It is a program that you install on your computer or mobile device, which contains the Softphone function and several other features and provides an interface for you to communicate. These options allow any computer to be used as a phone through headphones with a microphone or sound card.
Any computing device like Desktop (Windows, Mac, Linux), Tablet (Android, iOS), Smartphone (Android, iOS) can all run Softphone programs, providing a lot of options to choose from.
According to the SIP standard, several special high-end phones can support voice conference solutions (Audio Conference) and video conference solutions (Video Conference). Softphones can also help these solutions. Besides, a broadband connection and connection to a VOIP provider or SIP server is required.
How Does SIP Calling Work
SIP calling works based on SIP trunking. This is a method of providing telephone service to businesses that have their IP PBX.
It replaces the standard PRI line between organizations and traditional telephone carriers. A SIP link is a virtual connection between your business and ITSP (Internet Telephony Service Provider).
The structure of SIP is the same as client-server protocol – HTTP; before requests are sent to the server, they will be made by client machines. The server processes requests and sends a response back to the client. A request and response are transmitted to The IP has the INVITE and ACK messages. The IP makes the minimum assumption based on the transport protocol.
This protocol itself provides reliability and does not depend on it. TCP.SIP relies on Session Description Protocol (SDP) for outlining agreements for codec verification. SIP supports the description of sessions that allow participants to agree to set types of compatible media.
A SIP call works as follows:
First, the caller sends an invitation, and the called machine returns a 100 – Try response. Whenever the other side of the line starts ringing, a Ring-response (180) will proceed. 200 – OK and ACK- reception are the two commands in communicating progress.
The actual call is now transmitted as data over the RTP. When the caller hangs up, a BYE request will be sent and responded with 200 – OK.
The provider is only responsible for the connection for you to use, and the user self-administering the switchboard means that your team decides which features to activate and manage.
What are The Features of SIP calling
Using SIP, people around the world can communicate using their computers and mobile devices over the Internet. This is an important part of Internet Telephony and allows you to harness the benefits of VoIP (voice over IP) and rich communication experiences. Users can also register their current location with their home server.
Secondly, since SIP is IP based, it will minimize the consumption of both data networks and telephony. You can enjoy a centralized system with a variety of easily scalable digital streaming abilities and no physical infrastructure required, meaning no maintenance costs or Hardware.
The SIP trunk can directly connect to a selected internet telephone service provider – ITSP through the removal of a PSTN gateway. This will eliminate subscription fees and giving you more flexibility in extending your telecommunications service by offering more bandwidth boost options at a lower price point.
SIP trunking services tend to be much more flexible and mobile than older phone systems during a disaster. Most trunking services will take measures to ensure you can still make a call even when there is a network error or natural hurricanes and storms
What’s the Difference Between SIP and VoIP?
VoIP (Voice over Internet Protocol) is a term used to describe telecommunications signal broadcasting technologies based on the Internet. Meanwhile, SIP is one of the protocols used to implement VoIP, namely establishing and terminating audio and video calling sessions.
If VoIP is a collection of calls made over the Internet, SIP is one of the technical standards for devices to connect and interact with each other in a VoIP conversation.
That means you can make any call over the Internet, otherwise known as VoIP, and SIP is just one of the many protocols that are currently the most widely used. In other words, you can do VoIP with other protocols, but most businesses today will use SIP because it’s pretty standard.
However, many businesses today prefer a combination of VoIP and SIP, even though it does not optimize business costs. Looking further, when your business phone system grows to a certain extent and needs to promote multimedia voice channels such as video calls for meetings and conferences, you will need SIP as an interface standard method for connecting devices.
At that time, the combination of VoIP and SIP will help businesses manage the call system more closely and effectively.
See also: What Is SIP Trunking?
SIP is a very important protocol for a virtual PBX system. It is the thread that connects and exchanges calls between Hardware. SIP calling helps people around thư world to communicate using the Internet. We hope that you are now equipped with enough information about the topic “What is SIP calling?”.
Thank you for reading.