VOIP means that voice is transmitted in packets on the IP network using the IP protocol. With the VOIP protocol, voice communication is possible whether it is the Internet, an intranet or a local area network. In a network using VOIP, voice signals are digitized, compressed and converted into IP packets, which are then transmitted over the IP network. The VOIP signaling protocol is used to establish and cancel calls, transmitting the information needed to locate users and negotiate capabilities. The main features of the telephone network are low cost; the synthesis of data, voice and video on the same network; new services on a centralized network and simplified management of end users.

Currently, there are some VOIP protocol stacks that originate from various standards bodies and providers such as H.323, SIP, MEGACO, and MGCP.

H.323 is an ITU-T standard originally used for multimedia conferencing on a local area network (LAN) and later extended to cover VOIP. The standard includes both point-to-point and multipoint conferences. H.323 defines four logical components: terminals, gateways, gatekeepers, and multipoint control units (MCUs). Terminals, gateways, and MCUs are all considered termination points.

Session Initiation Protocol (SIP) is an IETF standard for establishing VOIP connections. SIP is an application layer control protocol used to create, modify, and terminate sessions with one or more participants. The structure of SIP is similar to HTTP (Client-Server Protocol). The client makes the request and sends it to the server, which sends a response to the client after processing the request. The request and response form a transaction.

The Media Gateway Control Protocol (MGCP) is a VOIP protocol proposed by Cisco and Telcordia that defines the communication services between the Call Control Unit (Call Agent or Media Gateway) and the Telephony Gateway. The MGCP is a control protocol that allows the central console to monitor IP phone and gateway events and notify them to send content to the specified address. In the MGCP architecture, intelligent call control is placed outside the gateway and handled by the call control unit (call agent). At the same time, the call control units keep in sync with each other and send a consistent command to the gateway.

The Media Gateway Control Protocol (Megaco) is the result of a joint effort between the IETF and ITU-T (ITU-T Recommendation H.248). Megaco/H.248 is a protocol for controlling the protocol elements of physically separate multimedia gateways so that call control can be separated from media conversion. Megaco/H.248 illustrates the link between a Media Gateway (MG) for converting circuit-switched voice to packet-based traffic and a Media Gateway Controller for service logic specifying such traffic. Megaco/H.248 informs the MG to connect data streams from outside the data packet or unit data network to a packet or unit data stream, such as Real Time Transport Protocol (RTP). From the perspective of VOIP structure and gateway control, Megaco/H.248 is quite similar in nature to MGCP, but Megaco/H.248 supports a wider network, such as ATM.

VOIP

— Voice over IP (also known as IP telephony, VoIP and digital telephony) — The path through which voice conversations are transmitted over a computer or other IP-based network.

SIP

— Session Initiation Protocol — An interactive user protocol developed by the IETF's MMUSIC working group and applied to standard initialization, modification, and termination of multimedia components such as video, voice, real-time messaging, online gaming, and virtual reality.

PSTN

– Public switched telephone network – a collection of world public lines – an exchange telephone network, like the Internet, is the hub of the world's public IP telephony packet transmission.

ISDN

- Integrated Services Digital Network - A type of VoIP switching line system that transmits digital (as opposed to analog) voice and data over ordinary telephone copper to achieve better voice quality and smoother transmission speeds than analog systems.

PBX

— Inter-enterprise telephone switching system (also known as enterprise communication switching system))))) - A telephone transmission system owned by a private enterprise, which is opposed to a telephone transmission system owned by a public carrier or a telephone company.

IVR

- In the field of telephony, interactive voice response - means that the computerized system allows individuals, such as phone callers, to follow the options in the voice menu, in other words, to interact with the computer system.

DID

- Direct dial-in (also known as direct dial-in in Europe) is a client component provided by the telephone company for the communication exchange system between enterprises. The telephone company (telecom) collects all the communication switches (PBX) connected to the customer. Number.

RFC

— Request description (in the form of RFCs) is one of a series of widely accepted web digital information files and standards in commercial and free software for the Internet and Unix communities.

Voice Quality - Since IP is used to transmit data, it does not provide real-time guarantees and only provides the most efficient service. In order for the user to accept voice communication over IP, the packet delay needs to be less than a limit.

Interoperability - In a public network environment, products from different providers need to operate with each other to promote a wider range of applications for VOIP.

Security - Protects VOIP signaling and control traffic with encryption (such as SSL) and tunneling (L2TP) technology.

Integration of the Public Switched Telephone Network (PSTN) - Internet telephony has been introduced, but it needs to work with the PSTN for the foreseeable future. Gateway technology is used to connect the two networks.

Scalability - VOIP needs to be flexible enough to grow with the growing private and public user market. To solve the above problems, many network management and user management technologies are gradually being formed.

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