Key Technology Analysis of IP router technology and IP Phone

IP router technology is still quite common, So I studied the Key Technology Analysis of IP router technology and IP Phone. Here I will share it with you and hope it will be useful to you. IP phones are the economic advantage of combining voice data integration with the technical progress of voice/grouped IP routers, thus usher in a new network environment, this new environment provides advantages such as low cost, high flexibility, high productivity, and enhanced efficiency. These advantages of IP Phones enable enterprises, service providers, and telecom operators to see many bright prospects. The opportunity to integrate voice and data into a group exchange network is driven by the following factors:

(1) Improve the efficiency through statistics multiplexing.
(2) improved efficiency through enhanced features such as voice compression and quiet suppression of voice activity detection.
(3) Long-distance cost is saved by transferring telephone calls over a private data network.
(4) reduce management costs through joint infrastructure components.
(5) The possibility of using new applications of computer telephone integration.
(6) voice connections on data applications.
(7) use the new IP router technology effectively.

The improved efficiency of grouping networks and the ability to statistically group multiplexing voice data streams with data groups allow companies to maximize the return on investment in data network infrastructure. However, putting voice data streams on the data network also reduces the number of dedicated voice lines, which are often very expensive. In the LAN, MAN, and WAN environments, new IP router technologies such as Kyrgyzstan Ethernet, dense wavelength division multiplexing, and PacketoverSDH are implemented to increase bandwidth at a lower price. Similarly, these IP router technologies provide better cost effectiveness than standard TDM connections.

IP Phone type

There are four types of IP Phones: phone to phone, phone to PC, PC to phone and PC to PC. The details are as follows:

1) PC-to-PC: At first, the IP address telephone method was mainly PC-to-PC. The IP address was used for call, and the voice compression and packaging transmission methods were used to realize real-time voice transmission between PCs on the Internet, 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.

2) telephone to telephone: A general telephone is connected to an IP Phone gateway through a telephone switch, and a call is made through the IP network using a telephone number. The sender gateway identifies the caller, translate the phone number/gateway IP address, initiate an IP phone call, connect to the gateway closest to the called device, and complete the voice encoding and packaging. The Receiving Terminal gateway can unpack, decode, and connect to the called device.

3) telephone to PC: the gateway is used to correspond to and translate IP addresses and phone numbers, as well as voice codec and packaging.

4) PC-to-Phone: the gateway is used to correspond to and translate IP addresses and phone numbers, as well as voice codec and packaging.

Key Technologies of IP Phones

Traditional IP networks are mainly used to transmit data services and adopt the best-effort, connectionless IP router technology. 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 IP phones include signaling, encoding, real-time transmission, and QoS.

1. signaling technology

Signaling technology ensures the smooth implementation of telephone calls and voice quality. Currently, the widely accepted VoIP control signaling system includes the H.323 series of ITU-T and the Session Initiation Protocol SIP of IETF.

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 IP router technical support for multimedia on the LAN, Wan, INTRANET, and Internet. H.323 is a set of protocols for ITU-T-related multimedia communications, including H.320 for ISDN, 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.225.0 for connection establishment, H.332 for large conferences, H.225.0 for service supplement, h.2.1, h.2.2, and h.2.3 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 recommended H.225.0 method is used for packaging and transmission of information flows.

Three types of signaling are involved in H.323 call establishment: RAS signaling R = Registration: Registration, A = license: Admission and S = Status: Status), H.225.0 call signaling, and H.245 control signaling. The RAS signaling is used to complete the registration, authorization, bandwidth change, status, and disconnection between the terminal and the network guard. The H.225.0 call signaling is used to establish the connection between the 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.

Although H.323 provides all the sub-protocols required for narrowband multimedia communication, the control protocol of H.323 is very complex. In addition, H.323 does not support multi-point transmission Multicast protocol. It can only use multi-point control unit (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. In contrast to H.323, SIP is a simple Session Initialization Protocol. Unlike H.323, which provides all communication protocols, it only provides session or call establishment and control functions. SIP can be used in Multimedia conferences, distance learning, Internet calls, and other fields. SIP supports both single-point Unicast and multi-point sending. session participants and media types can join an existing meeting at any time. SIP can be used to call people or machine devices, such as calling a media storage device to record a meeting, or calling an on-demand TV server to play a video signal to a meeting.

SIP is an application layer protocol that can use UDP or TCP as its transmission protocol. Different from H.323, SIP is a text-based protocol that describes SIPUniformResourceLocators using the SIP rule resource locating language. This makes it easy to implement and debug, and more importantly, it provides high flexibility and scalability. Because SIP is only used for initial calls, rather than transmitting media data, additional transmission costs are not high. The sip urll can even be embedded in web pages or other hypertext links. You only need to use a mouse to make a call. Compared with H.323, SIP also features fast call establishment and support for mobile phone numbers.

2. Coding Technology

Voice compression and encoding technology is an important part of IP phone technology. At present, the main coding techniques include G.729, G.723 (G.723.1) defined by ITU-T. 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 is extended to 6.4 ~ 11.8 kbit/s, the voice quality has also changed in this range, but even 6.4 kbit/s, the voice quality is also good, so it is very suitable for use in VoIP systems. G723.1 adopts 5.3/6.3 Kbit/s dual-rate voice encoding. Its voice quality is good, but the processing latency is large. It is the currently standardized lowest-rate voice encoding algorithm.

3. Real-Time Transmission Technology

The real-time transmission IP router 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 tag and a mechanism for controlling synchronization characteristics of different data streams. It allows the receiver to reorganize the data packets at the sending end and provides service quality feedback from the receiver to the Multi-Point sending group.

4. QoS) Assurance Technology

The IP Phone mainly uses the Resource Reservation Protocol (RSVP) and the real-time transmission control protocol RTCP for service quality monitoring to avoid network congestion and ensure the quality of calls.

5. Network Transmission Technology

In the IP phone, the network transmission IP router technology mainly includes 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, RTP can be used in IP phones 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 IP Phone format, multiple voices are inserted into the voice data segment in this way, which improves the transmission efficiency. In addition, the technology of voice detection and echo elimination is also a key IP router technology in IP phones. 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.