November 29, 2023

PBX Science

VoIP & PBX, Networking, DIY, Computers.

VoIP Realization Principle and Key Technology

8 min read

VoIP Realization Principle and Key Technology

VoIP Realization Principle and Key Technology


1. Basic principle and implementation form of VoIP


The IP telephone system converts the analog signal of an ordinary telephone into an IP data packet that can be connected to the Internet by a computer, and at the same time converts the received IP data packet into an analog electrical signal of sound.

After the conversion and compression processing of the IP telephone system, the transmission rate of each ordinary telephone occupies about 8-11kbit/s bandwidth, so when using the same bandwidth as the ordinary telecommunication network with a transmission rate of 64kbit/s, the number of IP telephones is the original 5-8 times.

The core and key equipment of VoIP is the IP telephone gateway.

The IP telephone gateway has a routing management function, and it maps the telephone area codes of each region to the corresponding regional gateway IP address.

The information is stored in a database, and the relevant processing software completes functions such as call processing, digital voice packaging, and routing management.

When a user makes an IP call, the IP telephone gateway determines the IP address of the corresponding gateway according to the data in the telephone area code database, and adds this IP address to the IP data packet, and selects the best route at the same time to reduce the transmission delay. The Internet reaches the destination IP telephony gateway.

For areas where the Internet has not been extended or no gateway has been set up temporarily, routing can be set up, and the nearest gateway can be switched through the long-distance telephone network to realize communication services.



VoIP Realization Principle and Key Technology


At present, the VoIP system is generally composed of IP telephone terminals, Gateway, Gatekeeper, network management system, billing system and other parts.

IP telephone terminals include traditional voice telephones, PCs, IP telephones, and multimedia service terminals integrating voice, data and images.

Since the data source structures generated by different types of terminals are different, to be transmitted on the same network, data conversion must be performed by a gateway or an adapter to form a unified IP data packet.

The IP telephone gateway provides the interface between the IP network and the telephone network. The user connects to the gateway of the IP network through the PSTN local loop.

The gateway is responsible for converting the analog signal into a digital signal and compressing it into an IP packet voice that can be transmitted on the Internet.

The signal is then transmitted to the gateway of the called user through the Internet, and the gateway of the called end unpacks, decompresses and decodes the IP data packet, restores it to an identifiable analog voice signal, and then transmits it to the called party’s gateway through the PSTN terminal. In this way, a complete phone-to-phone IP phone communication process is completed.

The gatekeeper is actually the intelligent hub of the IP telephone network and the service platform of the entire system, responsible for the management, configuration and maintenance of the system.

The functions provided by the gatekeeper include dial plan management, security management, centralized account management, database management and backup, network management and so on.

The function of the network management system is to manage the entire IP telephone system, including equipment control and configuration, data allocation, dial plan management, load balancing, and remote monitoring.

The function of the billing system is to calculate the cost of the user’s call and provide corresponding documents and statistical reports.

The billing system can be provided by the IP telephone system manufacturer, or it can be produced by a third party, but at this time, the IP telephone system manufacturer needs to provide its software data interface.


In terms of implementation, VoIP has four modes: telephone to telephone, telephone to PC, PC to telephone, and PC to PC.

Initially, the VoIP method was mainly from PC to PC, using IP addresses to make calls, and through voice compression and packet transmission, real-time voice transmission between PCs on the Internet was realized.

Voice compression, codec, and packaging were all through the processor and sound card on the PC. , network card and other hardware resources, this method is very different from public telephone communication, and is limited to the Internet, so it has great limitations.

Call-to-telephone means that the ordinary telephone is connected to the IP telephone gateway through the telephone exchange, and the telephone number is used to make a call through the IP network.

The sending gateway identifies the calling user, translates the telephone number/gateway IP address, initiates an IP telephone call, and connects to the nearest The called gateway completes voice encoding and packaging, and the receiving gateway implements unpacking, decoding and connection to the called party. For the case of phone to PC or PC to phone, the correspondence and translation of IP address and phone number, as well as voice codec and packaging are completed by the gateway.



2. Key technologies of VoIP

Traditional IP networks are mainly used to transmit data services, using best-effort and connectionless technologies, so there is no quality of service guarantee, and there are situations such as packet loss, out-of-sequence arrival, and delay jitter.

Data services do not have high requirements for this, but voice is a real-time service, which has strict requirements on timing and delay.

Therefore special measures must be taken to guarantee a certain quality of service.

The key technologies of VoIP include signaling technology, coding technology, real-time transmission technology, quality of service (QOS) guarantee technology, and network transmission technology.


2.1 Signaling Technology

Signaling technology ensures the smooth implementation of telephone calls and voice quality. Currently, the widely accepted VoIP control signaling system includes ITU-T’s H.323 series (used by Huawei products) and IETF’s session initiation protocol SIP.

The H.323 series of ITU recommendations define protocols and procedures for multimedia communication on the Internet or other packet networks without service quality assurance.

The H.323 standard is the technical foundation guarantee for multimedia on LAN, WAN, Intranet and Internet.

H.323 is a protocol set related to multimedia communication by ITU-T, including H.320 for ISND, H.321 for B-ISDN and H.324 for PSTN terminals.

Its encoding mechanism, protocol range and basic operation are similar to the simplified version of ISDN’s Q.931 signaling protocol, and adopt a more traditional circuit switching method.

Related protocols include H.245 for control, H.225 for connection establishment, H.332 for large conferences, H.450.1, H.450.2 and H.450.3 for supplementary services, security-related H.235, H.246 interoperable with circuit switching services, etc.

H.323 provides interoperability between devices, between high-level applications and between providers.

It does not depend on the network structure, is independent of the operating system and hardware platform, and supports multipoint function, multicast and bandwidth management.

H.323 has considerable flexibility and supports conferences between nodes with different functions and conferences between different networks.

he information flow in the multimedia conferencing system suggested by H.323 includes audio, video, data and control information.

The information flow adopts the H.225 recommendation method to package and transmit.

The H.323 call establishment process involves three kinds of signaling: RAS (RegistrationAdmissionStatus) signaling, H.225 call signaling and H.245 control signaling.

RAS signaling is used to complete the processes of registration, authorization, bandwidth change, status and disengagement between the terminal and the gatekeeper;

H.225 call signaling is used to establish a connection between two terminals. This signaling uses Q.931 messages to control the establishment and teardown of calls.

When there is no gatekeeper in the system, the call signaling channel open between two terminals; when a gatekeeper is included in the system, the gatekeeper decides to open a call signaling channel between the terminal and the gatekeeper or between two terminals;

H.245 control signaling is used to transmit terminal-to-terminal control messages, including master-slave discrimination, capability exchange, opening and closing logical channels, mode parameter requests, flow control messages, and general commands and instructions.

The H.245 control signaling channel is established between two terminals, or between a terminal and a gatekeeper.

In addition, H.323 does not support multicast (Multicast) protocol, and can only use a multipoint control unit (MCU) to form a multipoint conference, so it can only support limited multipoint users at the same time. H.323 also does not support call transfer, and it takes a long time to establish a call.


2.2 Encoding technology

Voice compression coding technology is an important part of IP telephony technology.

At present, the main coding technologies include G.729 and G.723 defined by ITU-T.

Among them, G.729 can compress the sampled 64Kbit/s voice to 8Kbit/s with almost no distortion quality.

Because the quality of service cannot be well guaranteed in the packet switching network, it is necessary to have a certain degree of flexibility in the coding of the voice, that is, the variable adaptability of the coding rate and the coding scale.

G.729 was originally the voice coding standard of 8Kbit/s, and now the working range is extended to 6.4-11.8Kbit/s, and the voice quality also has some changes within this range, but even at 6.4Kbit/s, the voice quality is still Not bad, so it is very suitable for use in VoIP systems.

G.723.1 adopts 5.3/6.3kbit/s dual-rate speech coding, its speech quality is good, but the processing delay is relatively large, it is the speech coding algorithm of the lowest rate that has been standardized at present.


In addition, silence detection technology and echo cancellation technology are also very critical technologies in VoIP.

Silence detection technology can effectively eliminate silent signals, thereby further reducing the occupied bandwidth of voice signals to about 3.5kbit/s; echo cancellation technology mainly uses digital filter technology to eliminate echo interference that greatly affects call quality, ensuring call quality.

This point is particularly important in IP packet networks with relatively large delays.


2.3 Real-time transmission technology

The real-time transmission technology mainly adopts the real-time transmission protocol RTP. RTP is a protocol that provides end-to-end real-time data transmission including audio.

RTP includes data and control two parts, the latter is called RTCP.

RTP provides a time stamp and a mechanism to control the synchronization characteristics of different data streams, allowing the receiving end to reassemble the data packets of the sending end, and can provide the quality of service packet feed from the receiving end to the multipoint sending group.


2.4 QOS guarantee technology

VoIP mainly uses Resource Reservation Protocol (RSVP) and Real-Time Transmission Control Protocol RTCP for quality of service monitoring to avoid network congestion and ensure call quality.


2.5 Network transmission technology

The network transmission technology in VoIP is mainly TCP and UDP, and also includes gateway interconnection technology, routing technology, network management technology, security authentication and billing technology, etc.

Since the real-time transport protocol RTP provides real-time, end-to-end data transmission services, VoIP can use RTP to transmit voice data.

The RTP header contains the identifier, serial number, time stamp, and transmission monitoring of the loaded data. Usually, the RTP protocol data unit is carried by UDP packets, and in order to minimize the delay, the voice payload is usually very short.

IP, UDP, and RTP headers are all counted by minimum length. The VoIP voice group has a lot of overhead, and the VoIP format of the RTP protocol is adopted.

In this way, multiple voices are inserted into the voice data segment, which improves the transmission efficiency.





VoIP Realization Principle and Key Technology

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