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A quick list of SIP terms for CIOs, CTOs, tech admins and everyone else curious about SIP
SIP, in telephony, is the acronym of Session Initiation Protocol. It is a signalling protocol used for initiating, maintaining, modifying and terminating real-time telecommunication involving voice, video, text messaging, etc.
Being a telephony technology that works behind the lines and often as invisible signals, SIP remains a mystic term for many. Today, we break down the mysticism that surrounds it. Here is a complete glossary of SIP terms and how they work in the telephony realm:
Before we get under the skin of SIP and the various terms in the SIP realm, it is necessary to understand the various flavors of SIP. Any discussion on SIP is bound to bring up the terms SIP calling, SIP trunking and SIP forwarding.
Here is what makes each one different from the other:
The process of sending/receiving a VoIP call containing voice, text, or multimedia elements is referred to as SIP calling.
The method of SIP calling where a SIP “trunk” is placed over the internet connection for telecommunication is SIP trunking.
The act of forwarding or redirecting of incoming SIP calls to another SIP location or phone number is SIP forwarding.
SIP is a relatively new concept in the telephony world. If you are someone who is going to use or work closely with SIP, here are some terms that you must add to your tech vocabulary to feel right at home.
The various elements or URIs (uniform resource indicators) that help SIP in creating the telephony network.
Like URLs in websites, Uniform resource indicators (URIs) identify the communications resource which helps the user to initiate and maintain a session with the resource.
Also known as a Location, a SIP peer refers to the number(s) on the virtual phone system pertaining to a particular location created and managed by the user. The location/SIP peer would contain the routing instruction (IP address of the phone numbers).
User agent is the endpoint in a SIP network. It could be a smartphone, softphone, IP phone, laptop, tablet device or any other device capable of internet communication.
Types of user agents:
A proxy server does the task of taking a request from a UA and forwarding it another user. It is placed in between users and acts like a router that sends and receives the SIP requests/responses.
A SIP registrar server does the task of accepting new registrations from users agents. It authenticates the users by storing their URI and location in the database. The same database would be used by other servers in the same network to authenticate the users.
A redirect server receives requests from user agents and generates 3xxx reponses that redirect the requests to an alternate set of URIs. They can be compared to traffic controllers who ensure that the flow of requests within the network is quickly processed.
Location service enables the redirect server or proxy server to trace the callee’s location. To trace the location, it maintains a list of SIP addresses or IP addresses.
Gateways act as bridges that help the SIp network to connect to other networks using different protocols or technologies. It helps convert traditional phone signals from a PSTN to other connections like VoIP thus facilitating seamless SIP calling.
A Session Border Controller (SBC) is used to demarcate the flow of data between sessions of various networks. For example, a SBC can be used to segregate sessions/calls made by two departments within a corporate network. In telephony, SBC is used for various purposes like rate limiting, network traffic policing, resource allocation, etc.
From the call origin until the recipient location, there are several parts of a SIP message that work behind the scenes.
A complete Request-Response is referred to as a transaction. A SIP transaction is said to have been created when the SIP request sent by a SIP peer is responded to by another SIP peer with a SIP response.
A series of transactions between SIP peers is referred to as a dialog. Dialogs are the building blocks of sessions which create, modify and complete the SIP session.
The exchange of data between two SIP peers is referred to as a session. In telephony context, a call between two endpoints. This could be a voice call, video call or a multimedia file exchange. The devices used for the SIP session could be a smartphone, laptop, IP phone or any other device equipped for VoIP calls.
Any SIP transaction is initiated by a request made by a User Agent (UA) to the server. The request could be to establish a connection, modify or terminate a session. Each SIP request will be answered with SIP response(s).
There are two types of SIP request methods - core methods and extension methods.
Core methods - INVITE, BYE, REGISTER, CANCEL, ACK, OPTIONS.
Extension methods - SUBSCRIBE, NOTIFY, PUBLISH, REFER, INFO, UPDATE, PRACK, MESSAGE.
The result of a received request sent by the User Agent (UA) server to the proxy server or any other SIP entity that initiated the session is referred to as response. SIP has 6 response codes - 1xx to 5xx taken from the HTTP protocol and 6xx, a new class defined by SIP.
List of SIP response request codes:
Session initiation protocol (SIP) uses a variety of protocols that help in sending and receiving or voice, text and multimedia messages. A quick look at the protocols involved are as follows:
SIP is a form of signalling protocol. Signalling protocol helps to identify the method used for physical transmission of data and locating destination networks where the voice is meant to be delivered.
HTTP stands for Hypertext Transfer Protocol. It forms the backbone upon which voice, text and multimedia communication is transacted on the internet. As a signaling protocol, SIP uses several HTTP protocols that enables communication.
TCP stands for Transmission Control Protocol. It is a network standard which prescribes how a network connection will be established and how it will continue to work. TCP send/receives data packets for telecommunication. User datagram protocol (UDP) is an alternative communications protocol to TCP. UDP is used for low-latency and loss-tolerating connections like voice calls.
Every message sent across the web or mobile has a message header and a body. In SIP, SDP is the protocol which defines the message bodies for exchange of voice calls using SIP.
The OSI model is used to define how data is sent and received over a network. It works by breaking down the data transmission over seven layers. Each layer is entrusted with a specific task that enables the message to reach its intended destination. It was developed as a communication standard by the International Organization for Standardization (ISO) in 1983.
A VoIP protocol which provides standards for voice, video, and data conferencing communication that use packet networks on several devices including computers, telephony equipment, networks, etc. It forms the backbone of SIP’s working.
IETF stands for Internet Engineering Task Force. It is Standards body for Internet protocols and services and maintains all the RFCs (Request for Comments). The IETF was originally an activity supported by the US Federal Government. Since 1993 it is operating as an international membership-based non-profit organization.
A type of publication from the technology community which invites engineers, scientists and tech professionals to offer their views on a said technology. RFCs are published by bodies like IETF (Internet Engineering Task Force), the Internet Research Task Force (IRTF), the Internet Architecture Board (IAB) or other independent authors. SIP is implemented on RFC 3261.
Be it SIP trunking or SIP forwarding, there are certain common components that work within the system facilitating the smooth happening of a SIP call.
The process of authentication followed to ensure that only authorized proxy server or User Agents (UA) are granted access to connections/service/features. It follows a challenge-based mechanism where the sender is required to prove his or her identity before processing the message.
A mechanism by which SIP header field names are shortened and displayed in abbreviated form. This helps in transporting messages that are either too large for TCP/UDP to transfer.
DTMF stands for dual tone multi frequency, which is the signal that a phone device creates when a user presses the phone key. In SIP, the signals are used to transmit the voice signals or used as instructions for IVR keypresses.
An outbound proxy is used when all calls are to be routed only through an outbound proxy server. It acts like an internet firewall in a corporate environment where all traffic is controlled and monitored.
From phone devices to telephony connections, here is a short list of all the basic hardware or network requirements necessary to make SIP work.
VoIP stands for Voice over internet Protocol. It is an internet-based communication medium that converts analog phone signals into digital signals using an intelligent process called packet switching. Read more about VoIP.
IP phone is a telephony device that uses VoIP signals or the internet for telecommunication. It does not require a PSTN line or a PRI line for telephony connection. They have jacks for connecting ethernet cables instead of phone jacks. IP phones are also referred to as hard phones, softphones, soft clients, VoIP phones, etc. They act as SIP extensions on which VoIP calls can be sent/received.
PRI stands for Primary Rate Interface. It is a form of ISDN (Integrated Services Digital Network) line that powers traditional phone connections. A PRI line enables users for heavy-duty telephony usage, as much as sending/receiving 30 calls concurrently.
PSTN stands for Public Switched Telephone Network. It is the existing network used by most phones around the world. A PSTN connection if offered by carrier networks like AT&T, T-Mobile, Verizon, Sprint, etc.
A virtual PBX is a cloud-based PBX system. It does not require telephony hardware to handle inbound or outbound calls. A virtual PBX can also perform additional tasks like call forwarding, call recording, call transferring, and so on.
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