IP-based voice transmission (English: Voice over Internet Protocol, abbreviated as VoIP) is a voice call technology that uses Internet Protocol (IP) to achieve voice calls and multimedia conferences, that is, communicate via the Internet. Other informal names are IP telephony, Internet telephony, broadband telephony, and broadband phone service.
VoIP can be used for many Internet access devices, including VoIP phones, smart phones, and personal computers, to make calls and send text messages via cellular networks and Wi-Fi.
You can understand
1 Principle
2 Voice coding
3 words
4 Legal status
5 birth defects
6 Common protocols
7 VoIP related technical standards
8 Development History
9 Common VoIP
10 Calculation method
11 Policy
12 Features
13 Security
Many friends like to use online chat tools for voice chat. This voice is not transmitted through the traditional telephone network of the telecom operator, but through the Internet. This technology that converts voice into IP data packets, partly or entirely based on IP network transmission, is VoIP (Voice over IP, voice over IP) technology.
The basic principle of voIP is to compress the voice data through the voice compression algorithm, and then package these voice data according to the TCP/IP standard, send the data packets to the receiving place through the IP network, and then string these voice data packets together , After decompression processing, it is restored to the original voice signal, so as to achieve the purpose of voice transmission via the Internet. The core and key equipment of IP telephony is the IP gateway, which maps the area code of each regional telephone to the corresponding IP address of the regional gateway. This information is stored in a database, and the data connection processing software will complete functions such as call processing, digital voice packaging, and routing management. When the user makes a long-distance call, the gateway determines the IP address of the corresponding gateway according to the telephone area code database data, adds this IP address to the IP data packet, and selects the best route to reduce the transmission delay. The IP data packet arrives via the Internet The gateway of the destination. In some areas where the Internet has not yet extended to or has not yet established a gateway, routing can be set up, and the nearest gateway can be transferred through the long-distance telephone network to realize communication services.
VoIP is a technology that focuses on IP phones and launches corresponding value-added services
There are three main ways of VoIP:
Internet phone: A voice call based entirely on Internet transmission, usually a call between a PC and a PC.
IP phone interconnected with the public telephone network: Voice transmission is realized through broadband or dedicated IP network. The terminal can be a PC or a dedicated IP phone.
VoIP services of traditional telecom operators: Voice transmission through the backbone IP network of telecom operators. The services provided are still traditional telephone services, using traditional phone terminals. By using an IP phone card or adding an IP dial prefix before the dialed phone number, this uses the VoIP service provided by the telecom operator.
VoIP is relatively cheap. This is because VoIP phones are just an application on the Internet. Essentially, VoIP calls are no different from emails, instant messages, or web pages. They can all be transmitted between machines connected to the Internet. These machines can be computers, or wireless devices, such as mobile phones or handheld devices.
Why do some VoIP services charge money, but some are free? VoIP service can not only communicate with VoIP users, but also with telephone users, such as users using traditional fixed-line networks and wireless mobile phone networks. For this part of the call, the VoIP service provider must pay the call fee to the fixed-line network operator and the wireless communication operator. This part of the charge will be transferred to VoIP users. Calls between online VoIP users can be free.
What are the requirements for using VoIP? First, you need to connect to the Internet. This can be the most basic dial-up Internet connection or a more ideal broadband service. The faster the network connection, the better the VoIP call quality. For example, a high-speed broadband connection allows you to make calls while surfing the Internet. Secondly, VoIP software is needed. Users can choose a VoIP software to install on a desktop or laptop computer. Then the computer can make online calls. If the user wants to convert his home phone into a VoIP dial-up system, he needs the help of an adapter. The VoIP software can be pre-installed separately in a hardware device called an "analog telephone adapter". The analog telephone adapter is mainly installed between the home phone and the broadband modem.
Many standards organizations and industrial entities in the world have proposed many voice coding schemes. Including the International Telecommunication Union's G.711 (rate 64kbps), G.723.1 (rate 5.3kbps or 6.3kbps), G.729A (rate 8kbps) coding scheme.
Network communication is unstoppable, and various instant messaging software impacts the traditional telephone communication methods, which has also created many network upstarts. Many investors hope to get a share of network communication.
Internet telephony is widely used by enterprises and individuals in long-distance calls at home and abroad. Some Internet phone "talkers" are rapidly emerging in schools, residential areas, and industrial areas. Unlike the traditional "IP public phone supermarket" that emerged a few years ago, the operating cost of VoIP is lower and the barrier to entry is lower. As long as a computer, a broadband, a few telephones plus a billing software can be operated, the network company can provide technical support to the franchisee or operator. Because it is as convenient as usual to make calls, the Internet "Talk Bar" quickly occupied the market for low- and middle-income and migrant workers, and had an impact on traditional telephone services.
Because of the low cost and the vast market, VoIP has become a "great profit" investment project. However, because there is no policy support, there are still loopholes in operation and management. The development prospects are not clear, and investment needs to be cautious.
After several years of development, although Internet telephony is widely used, the Internet telephony business is still in a "grey area", which is not restricted or permitted by law, and the Internet telephony "Talk Bar" on the market does not have a business license. Experts remind that investors should not invest lightly, despite the rapid development of Internet telephony services because they are not supported and protected by policies.
Call quality is affected by the quality of the network
Unavailable during power failure
There is a gap between clarity and traditional fixed-line phones (under normal network conditions, there is no significant gap between the call sound quality and traditional phones)
There is a risk of being overheard and secretly recorded
You can change the number at will, which is easy to cause crime (must be allowed by the operator, users can not change the number at will)
Commonly used protocols (Control Protocol) such as H.323, SIP, MEGACO and MGCP.
H.323 is an ITU-T standard, originally used for multimedia conferences on a local area network (LAN), and later expanded to cover VoIP. The standard includes both point-to-point communication and multipoint conferences. H.323 defines four logical components: terminal, gateway, gatekeeper and multipoint control unit (MCU). Terminals, gateways and MCUs are all regarded as terminal 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 sends a request and sends it to the server, and the server sends a response to the client after processing these requests. The request and response form a transaction.
The Media Gateway Control Protocol (MGCP) defines the communication service between the call control unit (call agent or media gateway) and the telephone gateway. MGCP is a control protocol that allows the central console to monitor IP phone and gateway events and notify them to send content to a designated address. In the MGCP structure, intelligent call control is placed outside the gateway and processed by the call control unit (call agent). At the same time, the call control units keep synchronized with each other and send consistent commands to the gateway.
The Media Gateway Control Protocol (MEGACO) is the result of joint efforts of IETF and ITU-T (ITU-TH.248 recommendation). Megaco/H.248 is a protocol used to control the protocol unit of a physically separated multimedia gateway, so that call control can be separated from media conversion. Megaco/H.248 describes the connection between a media gateway (MG) that is used to convert circuit-switched voice to packet-based communication traffic and a media gateway controller that specifies the service logic for such traffic. Megaco/H.248 informs the media gateway to connect the data stream from outside the data packet or unit data network to the data packet or unit data stream, such as real-time transport protocol (RTP). From the perspective of the relationship between VoIP structure and gateway control, Megaco/H.248 is quite similar to MGCP in nature, but Megaco/H.248 supports a wider range of networks, such as ATM.
In order to carry out multimedia applications on existing communication networks, the International Telecommunication Union (ITU-T) has formulated the H.32x multimedia communication series of protocols. The following is a brief description of the main standards:
H.320, a standard for multimedia communication on narrowband videophone systems and terminals (N-ISDN);
H.321, the standard for multimedia communication on B-ISDN;
H.322, a standard for multimedia communication on a local area network with QoS guarantee;
H.323, a standard for multimedia communication on packet-switched networks without QoS guarantee;
H.324, a standard for multimedia communication on low-bit-rate communication terminals (PSTN and wireless networks).
Among the above-mentioned standards, the network defined by the H.323 standard is the most widely used, such as Ethernet, token network, and FDDI network. The application based on the H.323 standard has naturally become a hot spot in the market, so we will focus on H.323 below. Four main components are defined in the H.323 recommendation: namely, terminal, gateway, gateway management software (also called gatekeeper or gatekeeper) and multipoint control unit.
(1) Terminal--All terminals must support voice communication, and video and data communication capabilities are optional. All H.323 terminals must also support the H.245 standard, which is used to control channel usage and channel performance. H.323 stipulates the main parameters of the voice codec in voice communication as follows: ITU recommends voice bandwidth/KHz transmission bit rate/Kb/s compression algorithm comment G.711 3.4 56,64 PCM simple compression, applied to G in PSTN .728 3.4 16 LD-CELP has the same voice quality as G.711, applied to low bit rate transmission G.722 7 48,56,64 ADPCM voice quality is higher than G.711, applied to high bit rate transmission G.723.1G.723.0 3.4 6.35.3 LP-MLQ voice quality is acceptable, G.723.1 adopts G.729G.729A for VOIP forum 3.4 8 CS-ACELP has a delay lower than G.723.1 and voice quality is higher than G.723.1.
(2) Gateway (Gateway)-This is an optional part of the H.323 system. The gateway can transform the protocols, audio and video coding algorithms and control signals used by different systems to adapt to the intercommunication of system terminals. If the H.324 system based on PSTN and the H.320 system based on narrowband ISDN communicate with the H.323 system, a gateway needs to be configured;
(3) Gatekeeper-This is an optional component of the H.323 system, which is managed by software. It mainly has two functions: the first is the management of H.323 applications; the second is the management of terminal communication (such as call establishment, teardown, etc.) through the gateway. The administrator can perform functions such as address translation, bandwidth control, call authentication, call recording, user registration, and communication domain management through the gatekeeper. An H.323 communication domain can have multiple gateways, but only one gatekeeper can work.
(4) Multipoint Control Unit--MCU realizes multipoint communication on the IP network, and point-to-point communication is not required. Make the whole system form a star topology through MCU. The MCU contains two main components: the multipoint controller MC and the multipoint processor MP, or it does not include MP. The MC processes the H.245 control information between terminals and establishes a minimal common namer for audio and video processing. MC does not directly process any media information flow, but leaves it to MP for processing. MP mixes, switches and processes audio, video or data information.
There are two parallel architectures in the industry for IP phones. One is the ITU-TH.323 protocol introduced above, and the other is the SIP protocol (RFC2543) proposed by the Internet Engineering Task Force (IETF). The SIP protocol is more suitable for Intelligent terminal.
VoIP mobile phone is also called VoIP dual-mode mobile phone or IP mobile phone for short. It perfectly integrates GSM and WiFi, dual-mode standby at the same time, sharing user information. As an ordinary mobile phone, with it, users can use traditional G network services when there is no WiFi environment, including calling, texting, MMS, GPRS, etc.; users can use it to enjoy high-speed Internet surfing wherever there is a wireless network , IM chatting, sending and receiving e-mail and other trendy mobile technologies. More importantly, people can use voip dual-mode mobile phones to call ordinary phones and mobile phones at an ultra-low price through the Internet, and it is free to talk to each other through the Internet. , And does not need the support of the operator, and at the same time, 3-party calls can be realized simply through the mobile terminal. As a PDA mobile phone, the VOIP dual-mode mobile phone has many intelligent and thoughtful designs in use, including handwriting input, personal information management, reading and editing of common document formats, and shooting high-pixel digital pictures.
While enjoying the various conveniences that VoIP technology brings to our lives, let us also understand the development history of VoIP:
Early VoIP was realized through software on personal computers. At that time, VoIP phones could only make calls between PCs and PCs, and the quality of the calls was not good. They were only regarded as an application of the Internet.
Between 2000 and 2002, VoIP technology began to penetrate the telecommunications field. The telephone service based on IP network appeared. At this time, the development of VoIP is very rapid, and VoIP can already be realized between PC-phone, PC-PC, and phone-phone. At this time, a lot of VoIP telephone service providers appeared, and the voice quality of VoIP was also continuously improving.
Since 2003, VoIP has developed into broadband phones. As a commercial service, VoIP has begun to compete with traditional fixed telephones. The voice quality of VoIP is similar to or even higher than that of traditional fixed-line phones.
With the development of mobile Internet, VoIP technology has also developed new trends:
The convergence of VoIP technology and wireless networks has emerged. Wireless networks mainly include 3G (The 3rd Generation Mobile Communications), LTE (Long Term Evolution), and WLAN (Wireless Local Area Network, wireless local area network). Among them, the development of WLAN-based VoIP is particularly rapid. In wireless VoIP, the final access of the signal is to use WLAN, and the other parts still use the wired network for transmission.
There is a convergence of VoIP and P2P (Peer-to-Peer, end-to-end). P2P technology comprehensively utilizes scattered network resources, making the connection rate and voice quality of voice calls even more than traditional telephone networks.
Domestic VoIP
GBPhone
Feiyang Internet Phone
263ET
Shutter Internet Phone
OCALLo
Internet phone lovers
REDVIP
Elite Club, etc.
Internet phone abroad
skype:
vbuzzer
globe7
wengo
MINO
webcall
freecall
voipstunt
sparvoip
GIzmo5
The bandwidth calculation method of common VOIP encoding is related to the selected encoding method, but has nothing to do with which manufacturer's, the formula is as follows:
Bandwidth = packet length × number of packets per second
=Package length×(1/packaging cycle)
=(Ethernet header+IP header+UDP header+RTP header+payload)×(1/packaging cycle)
= (208bit +160bit+64bit+96bit + payload) × (1/packing period)
= (528bit + (packing period (second) × number of bits per second)) × (1/packing period)
= (528 / packing period) + bits per second
According to various encoding methods, we get:
G711: 20ms packing, bandwidth is (528/20 + 64) Kbit/s=90.4 Kbit/s
G729: 20ms packing, the bandwidth is (528/20 + 8) Kbit/s = 34.4 Kbit/s
G723: 5.3k, 30ms packing, the bandwidth is (528/30 + 5.3) Kbit/s=22.9 Kbit/s
The industry generally designs the network bandwidth according to the IP network bandwidth coefficient and Ethernet bandwidth coefficient provided in the following table:
Encoding and decoding technology Compression rate (Kbps) Packing period (ms) IP network bandwidth coefficient Ethernet bandwidth coefficient
G.711 a/u 64 20 1.25 1.41
G.729 a/b 8 20 0.38 0.54
G.723.1(5.3kbit/s) 5.3 30 0.27 0.37
G.723.1(6.3Kbit/s) 6.3 30 0.25 0.36
H.263(384Kbit/s) ≈384 10 6 6.2
Remarks: When using a certain encoding method, multiply 64K by the corresponding bandwidth coefficient to get the actual bandwidth occupied. Of course, if it is a relay interface, you also need to consider that signaling occupies a certain bandwidth, which is generally calculated at 2.5%.
Policy editor
On March 17, 2012, the Ministry of Industry and Information Technology issued the "Notice on Doing a Good Job in the Follow-up Work of Issuing 3G Licenses", which clearly stipulated the business scope of the three operators.
In this document issued to the three major telecom operators, there is still no sign of loosening the WIFI and VOIP services. All three operators are allowed to develop IP telephony services, but they are all limited to Phone-Phone telephone services.
Out of the consideration of protecting the operator's voice service income, the competent government department has been prohibiting the development of WIFI and VOIP. But with the advent of the 3G era, the three major telecom operators have deployed
WIFI hotspots across the country, and private VOIP services are also developing in full swing.
On January 8, 2013, the Ministry of Industry and Information Technology published the "Mobile Communication Resale Service Pilot Program" on its official website and began to solicit opinions.
On May 17, the Ministry of Industry and Information Technology announced the "Mobile Communication Resale Service Pilot Program", marking the coming of the era of virtual operations.
Featuresedit
1. No computer is required. It will not be like a computer because it cannot be used correctly due to a virus or operating system failure.
2. Suitable for guaranteeing long-term online. The key to fighting each other on the intranet is to stay online.
3. Encrypted communication.
4. Clear sound quality. The voice processing adopts the international advanced voice processing chip, which effectively solves the problems of echo cancellation, data packet loss, and network jitter. The sound quality is generally higher than that of a softphone.
5. The cost is low, and the number can be changed and displayed at will.
6. It can be monitored and recorded, and it supports functions such as three-way communication, secret language, and call forwarding. [1]
There are few examples of attacks on VoIP. Most of the VoIP phone systems are installed in enterprises, and the encryption technology of VoIP phones also uses common technologies.
The security of V0IP is achieved through VoIP encryption, that is, through voice data encryption. The currently used encryption methods are TLS and IPSec.
The first step to take is to implement an authentication mechanism in your local area network, which includes devices and users. If you buy some products similar to 802.1x, they are not sufficient, because most of the phones, printers, physical examination equipment, robots, and other devices do not support the requirements of 802.1x. [1]
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