VOIP implementation principle and key technologies

Implementation principles and key technologies of VOIP: basic principles and implementation forms IP telephone systems convert analog signals of common telephones into IP data packets sent by computers over the Internet, at the same time, it also converts the received IP packet into a sound analog electrical signal. After the conversion and compression of the IP Phone System, the transmission rate of each common phone is approximately 8-11 kbit/s band width. Therefore, when the same transmission rate as that of a common telecom network is 64 kbit/s, the number of IP phones is 5-8 times that of the original one. IP Telephone Gateway is the core and key device of VOIP. The IP Telephone Gateway has the routing management function, which maps the telephone area numbers of various regions to the IP addresses of the corresponding regional gateways. The information is stored in a database and related processing software provides call processing, digital voice packaging, route management, and other functions. When a user calls an IP address, the IP address Telephone Gateway determines the IP address of the corresponding Gateway Based on the telephone area code database, adds the IP address to the IP address data packet, and selects the best route, to reduce transmission latency, IP data packets are sent over the Internet to the destination IP Phone gateway. For areas where the Internet does not extend to or where no gateway is yet set up, you can set a route. The nearest gateway transfers the route through the long-distance telephone network to implement communication services. Currently, the VOIP system consists of several parts, including IP phone terminals, gateways, keeper, network management system, and billing system. IP phone terminals include traditional voice phones, PCs, and IP phones. They can also be multimedia service terminals integrating voice, data, and images. Different types of terminals have different data source structures. to transmit data on the same network, the gateway or an adapter converts data to form a unified IP data packet. The IP Telephone Gateway provides interfaces between the IP network and the telephone network. You can connect to the IP network gateway through the PSTN local loop. The Gateway is responsible for converting analog signals into digital signals and compressing and packaging them, it becomes an IP group voice signal that can be transmitted over the Internet and then transmitted to the gateway of the called user over the Internet. The Gateway of the called end unpacks, unpacks, and decodes IP packets, it can be restored to a recognized Analog voice signal and then transmitted to the terminal of the called party through PSTN. In this way, a complete communication process from the phone to the IP address of the phone is completed. The Network guard is actually an intelligent hub of the IP telephone network. It is a service platform of the entire system and is responsible for system management, configuration, and maintenance. The Network Guard provides functions such as dial-up solution management, security management, centralized account management, database management and backup, and network management. The function of the network management system is to manage the entire IP Phone System, including Device Control and configuration, data allocation, dial-up solution management, Server Load balancer, and remote monitoring. The billing system calculates the charges for users' calls and provides corresponding documents and statistical reports. The billing system can be provided by the IP phone system manufacturer or by a third party. However, the IP phone system manufacturer must provide its software data interface. In terms of implementation, there are four methods for VOIP: telephone to telephone, telephone to PC, PC to telephone, and PC to PC. Initially, the VOIP method was mainly from PC to PC, and the IP address was used for call. Through voice compression and packaging transmission, real-time voice transmission between PCs on the Internet was realized, voice compression, encoding/decoding, and packaging are all done through hardware resources such as processors, sound cards, and NICs on the PC. This method is very different from public telephone communication and is limited to the Internet, so there are great limitations. A call refers to a call that is connected to an IP Phone gateway through a telephone switch. A call is made through an IP network using a telephone number. The sender gateway identifies the caller and translates the phone number/gateway IP address, initiate an IP phone call, connect to the gateway closest to the called call, and complete the voice encoding and packaging. The Receiving Terminal gateway can unpack, decode, and connect to the called call. The gateway is used to correspond to and translate IP addresses and phone numbers, as well as voice coding/decoding and packaging. 2. Key Technologies of VOIP traditional IP networks are mainly used to transmit data services and adopt best-effort and connectionless technologies. Therefore, there is no service quality guarantee, packet Loss, out-of-order arrival, and latency jitter exist. Data Services do not have high requirements for this, but voice is a real-time service and has strict requirements on time sequence and latency. Therefore, special measures must be taken to ensure the service quality. Key Technologies of VOIP include signaling, encoding, real-time transmission, QOS, and network transmission. 2.1 signaling technology to ensure the smooth implementation of telephone calls and voice quality, currently widely accepted VOIP control signaling system including ITU-T H.323 series (Huawei products) and IETF Session Initialization Protocol SIP. ITU's H.323 series recommendations define protocols and procedures for multimedia communication over the Internet or other group networks without service quality assurance. The H.323 standard provides technical support for multimedia on the LAN, Wan, Intranet, and Internet. H.323 is a set of ITU-T protocols for multimedia communications, including H.320 for ISND, H.321 for B-ISDN and H.324 for PSTN terminals. The encoding mechanism, protocol scope and basic operations are similar to the simplified version of the Q.931 signaling protocol of ISDN, and the traditional circuit switching method is adopted. Related Protocols include H.245 for control, H.225 for connection establishment, H.332 for large conferences, h.2.1, h.2.2 and h.2.3 for business supplement, and H.235 for security, h.246. H.323 provides interoperability between devices, between high-level applications, and between providers. It is independent of the network structure and operating system and hardware platform. It supports multi-point functions, multicast and bandwidth management. H.323 is flexible and supports meetings between nodes with different functions and between different networks. Information Flows in multimedia conferencing systems recommended by H.323 include audio, video, data, and control information. The information flow is packaged and transmitted in H.225 mode. Three types of signaling are involved in H.323 call establishment: RAS (Registration Admission Status) signaling, H.225 call signaling, and H.245 control signaling. RAS signaling is used to complete the registration, authorization, bandwidth change, status, and disconnection between the terminal and the network guard. H.225 call signaling is used to establish connections between two terminals, this signaling uses the Q.931 message to control the establishment and removal of calls. When there is no network punctuality in the system, the call signaling channel opens between the two terminals involved in the call. When the system includes a network punctuality, the Network guard decides to open a call signaling channel between the terminal and the network guard or between two terminals. The H.245 control signaling is used to send control messages from the terminal to the terminal, it includes master-slave identification, capability switching, opening and disabling logical channels, mode parameter requests, flow control messages, and Common commands and commands. The H.245 control signaling channel is established between two terminals, or between one terminal and one network guard. In addition, H.323 does not support multi-point transmission (Multicast). It can only use multi-point control units (MCU) to form multi-point meetings. Therefore, it can only support limited multi-point users. H.323 does not support call transfer, and it takes a long time to establish a call. 2.2 encoding technology voice compression encoding technology is an important part of IP phone technology. At present, the main coding technology has ITU-T defined G.729, G.723 and so on. G.729 compresses the sampled 64 Kbit/s voice to 8 Kbit/s with almost no loss of quality. Because the service quality in the group switching network cannot be well guaranteed, the voice encoding must be flexible, that is, the variable encoding speed and the variable encoding scale. G.729 was originally the 8 Kbit/s voice encoding standard, and now the scope of work has been extended to 6.4-11.8 Kbit/s. The voice quality has also changed in this range, however, even if it is 6.4 Kbit/s, the voice quality is also good, so it is suitable for use in VOIP systems. G.723.1 adopts 5.3/6.3 kbit/s dual-Rate Speech Encoding. The quality of speech is good, but the processing latency is large. It is the currently standardized lowest-Rate Speech Encoding Algorithm. In addition, the mute detection and echo elimination technologies are also critical technologies in VOIP. The mute detection technology can effectively remove silent signals and further reduce the bandwidth occupied by voice signals to around 3.5 kbit/s; echo Cancellation Technology mainly uses digital filter technology to eliminate echo interference that has a great impact on the quality of calls and ensure the quality of calls. This is particularly important in IP group networks with relatively large latency. 2.3 real-time transmission technology mainly uses the real-time transmission protocol RTP. RTP is an end-to-end protocol for real-time data transmission, including audio. RTP consists of two parts: data and control. The latter is RTCP. RTP provides a time label and a mechanism for controlling synchronization characteristics of different data streams. It allows the receiving end to reorganize the data packets at the sending end and provides the quality of service packet feedback from the receiving end to the Multi-Point sending group. 2.4 QOS Assurance Technology VOIP mainly uses Resource Reservation Protocol (RSVP) and real-time transmission control protocol RTCP for service quality monitoring to avoid network congestion and ensure the quality of calls. 2.5 network transmission technology the network transmission technology in VOIP is mainly TCP and UDP. In addition, it also includes gateway interconnection technology, Route Selection technology, network management technology, security authentication and billing technology. Since real-time transmission protocol RTP provides end-to-end data transmission services with real-time characteristics, VOIP can be used to transmit voice data. The RTP Header contains the identifier, serial number, timestamp, and transmission monitoring of the loaded data. Generally, the RTP data unit is carried by UDP groups, and to minimize latency, the voice load is usually very short. IP, UDP, and RTP headers are all calculated based on the minimum length. The cost of VOIP Voice grouping is very high. Using the RTP protocol VOIP format, multiple voices are inserted into the voice data segment in this way, which improves the transmission efficiency.