1. Basic principle of IP phone
The voice signal is compressed and encoded by the voice compression algorithm, and then the voice data is packaged according to the TCP / IP standard, and the data packets are sent to the receiving place through the network; the receiving end strings these voice data packets, and after decoding and decompression, the original voice signal is recovered, so as to achieve the purpose of transmitting voice from the Internet.
2. IP phone workflow
(1) Digitalization of speech
This is the first step of IP telephony. If the user uses a computer, the digitization is carried out in the computer; if the user uses an analog telephone, the voice is transmitted to the switching device through the access network, and then the voice is digitized by using devices such as PCM.
(2) Data compression
Data compression is used to eliminate useless signals and compress the voice signals after digitization (if ISDN terminal is used and ISDN is used as access, the working process starts from this step). This step is divided into two stages: a system analyzes the digitized signal, determines whether the signal contains voice, noise or voice gap, and then discards the noise and voice gap signal, which requires the system to have the ability to judge voice, noise and voice gap signal, and discards the noise and voice gap signal to send out the voice. B use complex algorithm to compress the speech digital signal without signal. Codec is the key part of this process.
(3) Data packaging
After the signal (also known as data) is compressed, it needs to be packaged to add some protocol information. In the process of collecting voice data, it needs some storage time (also known as time delay), because a certain amount of voice data must be received before it is sent to IP network. In the process of signal coding and compression, it also needs a certain amount of time to store the data, which also produces a certain time delay. Protocol information is added to the packet in order to better ensure the completion of the data transmission process. For example, each packet needs to contain the address information of a destination, a sequence number of packets added to prevent packets from arriving at the destination in an unordered manner, and data verification information. Because IP protocol is designed for the interconnection of different networks, compared with private network, it contains many complex processes. It requires to encapsulate a packet into another packet, and the data in the process of transmission has to go through the process of repackaging, re addressing and re packaging.
(4) Unpacking and decompression
When each packet arrives at the destination host (gateway, server or user computer terminal), it is necessary to check the serial number of the packet and put it in the correct position, and then use a decompression algorithm to recover the original signal data as much as possible. At this time, clock synchronization and time delay processing technology is used to fill the vacancy caused by the processing process of the sender. Because each packet in the transmission process through a different route, so they arrive at the destination in a very different order, so the receiver will first arrive at the packet into the jitter memory after a period of time to wait for the packet, the length of waiting time depends on the network congestion.
(5) Voice recovery
At present, IP phone is mainly used on the Internet, which is a value-added service on the Internet. Because Internet is an open network, and its bandwidth is not wide enough, especially in the case of network deterioration, a large part of packets will be lost or delayed in the transmission process. These discarded, delayed and damaged packets are the fundamental reasons for the decline of voice quality. According to the traditional error correction mechanism of Internet, if the receiver receives the wrong data packet, it will discard it and request retransmission. Therefore, the final data received by the user is exactly the same as the original data sent. Because IP telephony is a time sensitive service, that is, real-time service, retransmission mechanism can not be used, and special error detection and error correction mechanism is needed to reconstruct the voice and fill the gap, which requires the receiver to store a certain amount of received voice data, and then use a complex algorithm to "guess" the content of the lost packet to generate new voice information, so as to improve the efficiency The quality of communication. Therefore, the voice heard by the receiver is not exactly the same as that spoken by the sender, and part of the information is "Reconstructed" by the IP phone system.
3. IP phone system has four basic components
Terminal, gateway, multipoint control unit (MCU) and gatekeeper.
(1) The terminal equipment is an IP phone client terminal, which can be software (such as IP phone of VocalTec company, NetMeeting of Microsoft company) or hardware (such as dedicated Internet phone). It can be directly connected to the IP network for real-time voice or multi-body communication.
(2) Gateway is the key equipment to provide PC to phone, phone to PC and phone to phone voice communication through IP network. It is the interface equipment between IP network and PSTN / ISDN / PBX network. It should have the following functions:
A. It has IP network interface and interface with PSTN / ISDN / PBX switch;
B complete the real-time speech compression, and compress the 64kbit / s speech signal into low bit rate speech signal;
C completes addressing and call control.
(3) The gateway is responsible for user registration and management, and mainly completes the following functions:
A address mapping: mapping e.165 address of telephone network to IP address of corresponding gateway;
B call authentication and management: to authenticate the identity of the access users and stop the access of illegal users;
C call record: make operators have detailed data to charge;
D area management: multiple gateways can be managed by one gateway.
(4) The function of multi-point access control unit (MCU) is to use IP network to realize multi-point communication, so that IP phone can support multi-point applications such as network conference. IP phone adopts gateway technology, one side of the gateway is connected to the traditional circuit switching network, such as PSTN, which can communicate with any external telephone; the other side of the gateway is connected to the packet switching network, such as Internet, intranet, extranet, etc.
In the whole IP phone system, the gateway is set up in all parts of the world to complete the access and conversion between local telephone network and Internet. After receiving the standard telephone signal, the gateway, after digitization, coding and compression, packs it to the Internet according to the IP protocol, and sends it to the opposite gateway through the Internet according to the transmission route. On the contrary, the gateway receives the IP packet from the Internet, decompresses it, restores it to the analog voice signal, and then transfers it to the telephone network system. The gateway can access and transfer voice signals at the same time to realize full duplex communication.
Gateway varies with different manufacturers, but the basic modules are the same, including data processing host, voice module, data processing module, data connection module and management software module. The gateway has the function of route management. It maps the area code of each area to the IP address of the corresponding area gateway. The information is stored in a database. Data connection processing software will complete call processing, digital voice packaging, routing management and other functions. When a user dials a long-distance call, the gateway determines the IP address of the corresponding gateway according to the area code database data, adds the IP address to the IP packet, and selects the best route to reduce the transmission delay. The IP packet arrives at the destination gateway through the Internet. In some areas where the Internet has not been extended or a gateway has not been set up, routing can be set up, and the nearest gateway can be used to transfer through the long-distance telephone network.
4. Key technology and application of IP phone
As we all know, the working principle of IP post time is to first carry out analog-to-digital conversion, coding, compression and packaging of the voice signal, and then transmit it through the Internet network, and then unpack, decompress, decode and digital to analog conversion to recover the voice signal. The key technologies related to IP phone call quality can be summarized into the following seven aspects.
(1) Speech compression technology. The technical foundation of IP phone is speech compression technology. As for the technical standard of speech compression, 64kbit / s PCM (pulse code modulation) was formulated in 1972; 32Kbit / s ADPCM (adaptive differential pulse code modulation) was formulated in 1986; 16kbit / s LD CELP (low delay code excitation linear prediction) was formulated in 1991; ITU (International Telecommunication Union) approved the speech compression standard in November 1995 after a long period of research Quasi-g.729, 8 kbit / s CS ACELP (linear prediction with symmetric structure excitation). At present, the standard for IP phone is G.723.1, and the rate is 5.3 / 6.3 kbit / s.
In addition, 4.8kbit/s, 2.4kbit/s and 1kbit/s speech compression technologies have been reported, but there is no international standard.
(2) Noise reduction technology. Also known as voice activation technology, it refers to the technology that stops sending voice packets when it detects the quiet period in the process of a call. A large number of studies have shown that only 36% - 40% of the signals are active or effective in a full duplex telephone conversation. When one party is speaking, the other party is listening, and there are a lot of significant pauses in the process of speaking. The network bandwidth can be greatly saved by the technology of noise reduction.
(3) Echo cancellation technology, in PBX or office switch side, a small amount of power is not fully converted and returns along the original path, forming echo. If the caller is not far away from the PBX or the office exchange, the echo will return quickly and the human ear can not hear it. However, when the echo return time exceeds 10 ms, the human ear can hear the obvious echo. In order to prevent echo, echo cancellation technology is generally used. Because the delay of general IP network can easily reach 40 ~ 50ms, echo cancellation technology is very important for IP telephone system.
(4) Speech jitter processing technology. One of the characteristics of IP network is network delay and network jitter, which can lead to a significant decline in the quality of IP calls. Network delay refers to the average transmission time of IP packets on the network, and network jitter refers to the change of transmission time of IP packets. When the voice delay on the network is more than 200ms, the two sides prefer to use half duplex communication. On the other hand, if the network jitter is serious, some voice packets will be discarded because they are late, which will produce intermittent voice and partial distortion, and seriously affect the voice quality. In order to prevent this kind of jitter, people use jitter buffering technology, that is, a buffer pool is set at the receiving end. When the voice packet arrives, it is buffered first, and then the system takes the voice packet out of the buffer pool at a stable and smooth rate and processes it, and then plays it to the receiver.
(5) Voice priority technology. Voice communication requires high real-time performance. In the IP network with insufficient bandwidth, voice priority technology is generally needed, that is, the highest priority of voice packet must be set in the IP network router. In this way, the impact of network delay and network jitter on voice will be significantly improved.
(6) IP packet segmentation technology. Sometimes there are long packets on the network, one packet is thousands of bytes. If such a long packet is not limited, it will also affect the voice quality in some cases. In order to ensure the call quality of IP phone, the size of IP packet should be limited to 2556 bytes.
(7) VoIP forward error correction technology. In order to ensure the voice quality, some advanced VOIP gateways use channel coding and interleaving technology. IP packets may be damaged or lost in the process of transmission. Using forward error correction technology can reduce the accumulation of error codes in the process of transmission. Of course, for the internal network with low packet loss and error rate, it is unnecessary to use this technology.
5. Talking mode of IP phone
Generally, IP phone calls are divided into PC to PC, PC to phone and phone to phone. Ethernet telephone is a kind of terminal based on H.323 protocol. It occupies an independent IP address and can directly access the network. It is a kind of special IP telephone, which can be directly connected to the Internet instead of PC. Due to the emergence of Ethernet telephone, IP telephone will be more widely used. Therefore, the terminal of IP telephone described below no longer refers to PC, but only refers to Ethernet telephone and ordinary telephone.
There are many ways to realize voice communication by using PSTN and Internet. ① It is the voice service in the pure PSTN network, which is related to IP phone; ② it is the voice service in the pure IP network, which is similar to the "PC to PC" mentioned above, but the terminals of both sides are Ethernet phones and directly connected to the Internet; ③ it is the call between Ethernet phones and ordinary phones, which is similar to the "PC to phone" mentioned above, but both sides can be the calling party; ④ it is the call between ordinary phones The call carried out by Internet transmission, i.e. "phone to phone", is also China Telecom. The mode of IP telephony of China Unicom, Jitong and Netcom; ⑤ refers to the communication between Ethernet phones through PSTN transmission, which may be applied to the communication between Ethernet phones of two places with PSTN connection but without Internet connection; it can also be the chain building process of getting the IP address of the called party through PSTN first, and then calling through Internet. It can be seen that ③, ④ and ⑤ are hybrid IP telephony services, which can ensure the interconnection between IP and PSTN by providing signaling adaptation, media control and media adaptation through IWF.
It must be noted that the connection on the Internet is based on the IP address. For example, the "mode 2" above is essentially the communication established by the Ethernet telephone through the Internet with the IP address as the telephone number. However, people are not used to the IP address for the telephone number, but more used to the traditional telephone number, which is another problem that must be solved.
PBX system based on IP Technology
At present, some of the above-mentioned seven key problems have been solved, some are being solved or will be solved in the near future, which makes IP telephony and various systems based on IP technology continue to emerge. Since 1999, the State Key Laboratory of mobile communication of Southeast University has cooperated with Hong Kong Communication Technology Center to design and develop a new generation of distributed PBX system based on IP technology with the support of the innovation and technology fund of the Hong Kong Special Administrative Region government. The system can effectively solve the demand of voice and data integration. It can make use of Internet to conduct voice communication and data communication independently, map IP address to telephone number in the system through system management, and connect with PSTN network through gateway. The system has good openness and scalability, and can meet a wider range of needs.
The block diagram of our PBX system based on IP technology is mainly composed of four parts: call manager, pbxhub, PBX gateway and Ethernet telephone.
Ethernet telephone occupies an independent IP address and can directly access the network. It mainly completes the following functions: A / D and D / a conversion of speech signal. Complete the compression and de compression of data signal, complete the packaging and unpacking of data according to H.323 protocol, and process the corresponding control and call signaling.
Pbxhub is actually an H.323 terminal, occupying an IP address. Considering the limited IP address resources and the compatibility with existing analog phones, pbxhub provides 8 ports of analog phones with independent IDs. It has one more interface module with analog telephone than Ethernet telephone, and other processes are the same.
PBX gateway mainly completes the conversion of signaling and protocol between IP network and PSTN network. One gateway provides 8 line interfaces to the central office. In addition to the hardware interface to the central office, its software must also support various telephone signaling standards.
Call manager is similar to the gateway in H.323 network, but it is actually a control center. In addition to the basic functions of gateway, it also has the functions of traditional PBX and more value-added functions. It is a software running on the NT server, which manages the resources of the whole system and realizes the functions of equipment management, call processing, user management, database management, billing management, system maintenance, etc. it is also through it that all kinds of IP phone calls mentioned above can be realized. At the same time, it also provides a powerful voice mail function to improve the quality of service and coordinate the user operation process.