Brief introduction of VOIP test content

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VoIP training materials 2
　　
Evolution of IP phone technology
　　
IP telephony has been developing rapidly since 1995 with its economic, high efficiency and technological development beyond the times. At present, it has become one of the most competitive technologies in data and voice communication. Many countries in the world have opened IP phone operation business, and China‘s IP phone trial operation has been more than half a year. IP technology is showing vigorous vitality, which will certainly promote the further development of the information industry. The development of IP phone has gone through two primary stages, and is now evolving to the third stage at a high speed
Unity and integration.
　　
1. Technology accumulation stage
　　
In the stage of technology accumulation, experts in CTI field put forward the idea of packet voice transmission: all packet voice systems follow a common mode, and packet voice transmission network can adopt IP, frame relay or ATM. At the edge of these networks, a device or component called "voice agent" is set up. Its task is to convert the voice information from the traditional voice format to the format suitable for packet transmission, and then send the packet data to the voice agent device at the destination through the above network.
　　
Two problems need to be solved in the voice proxy connection mode in the packet voice network transmission system to make the packet voice service meet the needs of users. The first is the conversion of speech coding, that is, how to convert speech information into digital signals; the other is signaling conversion, which mainly identifies the object that the caller calls and the location of the caller in the network.
　　
Human language is expressed in the form of analog signal. The early telephone analog signal can be described as a smooth "sine wave". Although the analog communication technology has been quite developed, the transmission efficiency is not high. When the transmission attenuation leads to the weakening of the analog signal, it is very difficult to distinguish the complex analog voice information from the transmission noise.
-- digital signal has only two states of "1" and "0", which is easy to distinguish from noise and is not easy to be distorted. Therefore, the global communication system has been converted to digital transmission format, called pulse code modulation (PCM), which converts analog voice to digital format. The standard telephone PCM uses 8-bit code and 8000 / s sampling frequency, so each telephone occupies 64KB / s channel bandwidth. Another telephone voice standard called adaptive differential PCM (ADPCM) converts voice into 4-bit code, so it only occupies 32KB / s. ADPCM is usually used for long-distance lines.
Based on this technology, the first generation of IP phone equipment has been successfully developed. By using the voice acquisition principle of sound card on computer, 64kbgs analog voice is converted into ADPCM digital signal, and the primary IP phone function from computer to computer is realized on I Internet. This kind of system mainly uses the computer to complete the speech compression and control, so it can only realize the real-time communication of one voice. For example, on the computer system of pii233, only four voice channels can be completed at most. In this gray system, there are many practical IP phone systems, such as the iPhone of VocalTec, the NetMeeting system of Microsoft, etc. The successful development of the first generation IP system has aroused people‘s great interest in IP telephone system, thus promoting the application research of IP telephone technology. People hope to use IP telephone system like general telephone system.
　　
2 practical stage
　　
- the second development stage of IP telephony is a leap forward on the basis of the first stage. It can not only use IP telephony system to communicate like PSTN system, but also realize the call with large traffic. Using the PSTN switching system, the stage of IP phone communication is called "practical stage". In the practical stage, IP phone is mainly a network access device, which completes data network transmission and PSTN transfer function. A practical IP telephone access terminal system (we call it gateway) generally consists of five parts
　　
-. Establish and control telephone connection, communication and disconnection
Speech compression and data coding
Data network transmission and control:
-. System maintenance
-. User information management
Such systems are still built on computer systems, but they are not end-user devices. Therefore, for general users, only one telephone is needed to realize IP communication. Next, we will study the functions and implementation methods of each part.
　　
2.1 establish and control telephone connection, communication and disconnection
　　
- establishing and controlling telephone connection, calling and disconnection is the information exchange interface between IP telephone system and PSTN, and is also the gateway for the current general telephone system to convert to Internet / Intranet. This part of the work is mainly through the telephone card (such as E1 card) programming control to achieve.
　　
Because E1 card can accept PSTN information, remove the relevant signaling and record it into pure digital voice signal, the signaling conversion is basically completed by E1 card. But in a complete IP telephony gateway, all parts must exchange information and coordinate with each other. Full information exchange is needed between E1 card and voice compression card, between voice compression card and NIC, and between components and user interface. The exchange of these information can be controlled by the behavior of state machine.
2.2 telephone connection and disconnection
　　
First, the caller a on the PSIN picks up the phone. After receiving the signal from the caller, the sender Capitel sends a dial tone or IVR (interactive voice response) prompt to the caller. When the caller hears the dial tone, he starts dialing and sends the called number to Capitel.
　　
- Capitel at end a selects the IP address and the best path according to the called number, and sends channel occupancy signal to Capitel at end B on the selected path, that is, the out signal of Capitel at end a occupies the in signal of Capitel at end B. Then Capitel at the a end sends the called number to Capitel at the B end. Note: this system takes the Capitel IP telephone system of Beijing post and telecommunication equipment factory as an example.
- Capitel at the B end converts the pure digital signal into PCM signal according to the called number and sends it to the PSTN at the B end to connect the called user. The called user answers and sends the off hook signal to the Capitel at the B end, and then the Capitel at the B end forwards it to the Capitel at the a end, and the two sides start to talk. When the call ends, if the a-end user hangs up first, the calling user sends the recovery or disconnection signal to Capitel, and the b-end Capitel sends the signal to the PSTN of b-end; if the b-end user hangs up first, B sends the recovery or disconnection signal to Capitel of a-end, and everything is recovered.
　　
-- 2.3 data processing of speech compression
　　
- speech compression is mainly used to compress speech signals. The commonly used speech processing methods include G.711, g.722, G.729 and G.723. These compression algorithms must be completed on hardware, otherwise, it is impossible to realize the call task with large traffic. This part can use the program to control the language compression card, so that it can process the speech signal in real time according to our needs. When the voice data acquisition is completed, it must be put into the memory. In the first step, the uncompressed digital signal must be collected, and then compressed and sent to the specified memory according to the required structure. Under the control of CPU, the DSP algorithm is used to compress the corresponding data. After compression, the voice signals are grouped and coded to form standard data packets, and then the data of these packets are sent to the network for transmission in the form of stream.
　　
-- 2.4 data coding processing
　　
Data encoding and processing is the main work of H.323 module. It is the key to the mutual reception of voice data transmission formats on different systems. The protocol was published by ITU on May 28, 1996. It has been widely used in multimedia data communication. It is a multimedia communication protocol used in Integrated Services Digital Network (ISDN).
Specific protocol standards include: h.255.0 (call processing protocol), H.245 (control processing protocol), H.261 and H.263 (video processing protocol), t.12o (data processing protocol). In the IP phone system, the main tasks are as follows
Real time audio coding
Control protocol
Data transmission protocol
　　
- 2.5 - data exchange between gateways
　　
Data exchange between gateways is a very important and difficult technology in the development of IP telephony system. Each manufacturer claims to meet the basic requirements of the IP and H.323 protocol. Compared with voicaltec, the founder of IP phone, and Capitel IP Switch System of Beijing post and telecommunication equipment factory, both products meet the specification of H.323, but the processing of G set in H.323 protocol is quite different, because the processing method of G set is not specified in H.323. VocalTec company adopts three-step coding method to package H.323 packet, while Capitel IP switch system adopts Chinese standard eight step coding method to package H.323 packet. In this way, when the two products communicate with each other, due to different packaging methods of H.323 packet and different explanations of received H.323 packet, incompatibility occurs.
　　
Serial number requirements
　　
The gateway supports the multimedia digital signal codec protocols of G.729A and G.723. G. 729a gives priority to G.723.1
Gateway 2 supports DTMF and MF decoding and encoding (when calling out), and finally IVR system can be used
The gateway supports the interworking with the gateway of the exchange center
"Fast setup" in 4-gateway support protocol H.323 V2
5. The end-to-end information record code can be passed between gateway and gateway
Gateway 6 can make use of the call confirmation from the settlement system and the operator of the settlement system for authentication
7. Detailed call records can be generated and delivered to the settlement center in real time
8 sales in line with iNO! For the gateway and gateway of version 2.0, we must first use ino! The authority of the organization, the successful completion of iNO! Organization‘s certification process
9 for the call of settlement center, through the signal of settlement center and terminal, iNO! Platform provides compatibility
10 provide the intercommunication function between gateway and settlement system
11 can transmit gateway routing call signaling and terminal routing call signaling
At least 24 hours, the local gateway clock can be synchronized with an accurate and reliable time source
Using the following algorithm, we can generate the terminal source code in the CDR of the settlement system: H.323 1000 Q.931
14 gateway supports T.38 fax protocol. Mandatory support for TCP / UDP / IP and V.21, v.27, V.17
Regarding the call of settlement system, the gateway will guarantee the integrity of the information
VOIP training materials 2

　　The evolution of IP telephony technology

　　——IP telephone has been developed rapidly since 1995 due to its economy, high efficiency and super-era technological development, and has now become one of the most competitive technologies in data and voice communication. Many countries around the world have opened IP phone operations, and my country’s IP phone trial operation has been working for more than half a year. IP technology is showing vigorous vitality and will definitely promote the further development of the information industry. The development of IP phones has gone through two primary stages. Stage, is currently evolving to the third stage at a high speed
　　-unified integration.

　　1 Technology accumulation stage

　　——In the technology accumulation stage, experts in the CTI field put forward the idea of ​​grouping voice transmission: all packet voice systems follow a common mode, and the packet voice transmission network can use IP, frame relay or ATM. Set up devices or components called "voice agents" at the edge of these networks, whose task is to convert voice information from a traditional voice format to a format suitable for packet transmission, and then send the packet data to the voice of the destination through the above-mentioned network On the proxy device.

　　——The voice proxy connection mode needs to solve two problems in the packet voice network transmission system, so that the packet voice service can meet the needs of users. The first is the conversion of voice coding, that is, how to convert voice information into digital signals; the other problem is signaling conversion, which is mainly to identify the object of the caller and the location of the caller in the network.

——Human language is expressed in the form of analog signals. Early analog telephone signals can be described as smooth "sine waves". Although analog communication technology has been quite developed, the transmission efficiency is not high. When transmission attenuation causes analog signals When it becomes weak, it is difficult to distinguish complex analog voice information from transmission noise.
　　——The digital signal has only two states of "1" and "0", which is easy to distinguish from noise and is not prone to distortion. Therefore, the global communication system has been converted to a digital transmission format, called pulse code modulation (PCM), which converts analog voice into a digital format. Standard telephone PCM uses 8-bit code and 8000/sec sampling frequency, so each telephone occupies 64kb/s channel bandwidth. Another telephone voice standard called adaptive differential P CM (ADPCM) converts voice into 4-bit code. So only occupy 32kb/s, ADPCM is usually used for long-distance lines.
——Based on this technology, people successfully developed the first generation of IP telephone equipment, using the sound card voice collection principle on the computer to convert the 64kb/s analog voice into ADPCM digital signal, and realize the computer to the computer on the Internet The primary IP phone function. Because this kind of system mainly uses the computer to finish the speech compression and control, so, generally can only realize the real-time communication of one way of speech. For example, on the computer system of PII233, only 4 voice channels can be completed at most. In this gray system, there are many practical IP phone systems, such as Vocaltec’s IPhone and Microsoft’s Netmeeting system. The successful development of the first-generation IP system has aroused people's great interest in IP telephone systems, and thus promoted the application research of IP telephone technology. People hope to use IP telephone systems like general telephone systems.

　　2 Practical stage

　　——The second development stage of IP telephony is a leap based on the first stage. It can not only realize the communication using the IP telephone system like the PSTN system, but also realize the call of large volume of traffic. Using the current P STN switching system, the stage of IP telephone communication is called the "practical stage." The IP phone in the practical stage is mainly a network access device, which completes the data network transmission and PSTN switching functions. A practical IP telephone access terminal system (we call it Gateway) generally includes five parts:

　　——. Establish and control the connection, call and disconnection of the phone
　　——. Voice compression and data encoding processing
　　——. Data network transmission and control:
　　——. System maintenance part
　　——. User Information Management
　　—This type of system is still built on a computer system, but it is not an end-user device. Therefore, for general users, only one telephone is needed to realize IP communication. Let's study the functions and implementation methods of each part.

　　2.1 Establish and control telephone connection, call and disconnection

　　——Establishing and controlling the connection, call and disconnection of the telephone is the information exchange interface between the IP telephone system and the PSTN, and is also the gateway for the conversion of the current general telephone system to the Internet/Intranet. This part of the work is mainly achieved through the programming control of the telephone card (such as E1 card).

　　——Because the E1 card can accept PSTN information, remove the related signaling, and record it into a pure digital voice signal. Therefore, the conversion of signaling is basically done by the E1 card. But in a complete IP telephone gateway, various components must exchange information with each other and coordinate their work. There needs to be sufficient information exchange between the E1 card and the voice compression card, between the voice compression card and the network card (NIC), and between various components and the user interface. The exchange of these information can be controlled by the behavior of the state machine.
2.2 Connection and disconnection of telephone

　　——First, the caller A on the PSIN picks up the phone. After receiving the caller's off-hook signal, the originating Capitel sends a dial tone or) IVR (Interactive Voice Response) prompt to the caller. The calling party hears the dial tone, starts dialing, and sends the called number to Capitel, the switch at end A.

　　——Capitel at end A selects the IP address and the best route according to the called number, and sends a channel occupation signal to Capitel at end B on the selected route, that is, the outgoing signal from Capitel at end A occupies the incoming signal from Capitel at end B. Then Capitel on the A side sends the called number to Capitel on the B side. (Note: This system takes the Capitel IP telephone system of Beijing Post and Telecommunications Equipment Factory as an example).
　　——Capitel at the B end converts the pure digital signal into a PCM signal according to the called number and sends it to the PSTN at the B end to connect to the called user. The called user picks up the phone to answer, and sends the off-hook signal to Capitel at the B end, and then the Capitel at the B end forwards it to the Capitel at the A end. When the call ends, if the user at end A hangs up first, the calling user sends a recovery or disconnect signal to Capitel, and Capitel at end B sends this signal to the PSTN at end B; if the user at end B hangs up first, then B to A Capitel sends a recovery or disconnection signal, and everything is recovered.

　　——2.3 Data processing of voice compression

——Voice compression mainly compresses the voice signal. Commonly used voice processing methods are: G.711, G.722, G.729 and G.723. These compression algorithms must be processed on hardware, otherwise, they will not It is possible to achieve a call task with a large volume of traffic. This part can use the program to control the language compression card, so that it can process the voice signal in real time according to our needs. When the voice data collection is completed, it must be put in the memory, in the process of collection. The first step must be to collect uncompressed digital signals, and then after compression processing, send them to the designated memory according to the required structure, and under the control of the CPU, use the algorithm in the DSP to perform the corresponding data compression processing. The compressed voice signal is then grouped and coded to form standard data packets, and then these packetized data are sent to the network for transmission in the form of streams.

　　——2.4 Data encoding processing

——Data encoding processing is the main work to be completed by the H.323 module. It is the key to whether the sending format of voice data can be mutually received on a mutually different system. The agreement was announced by the ITU on May 28, 1996. It has been widely used in multimedia data communication. It is a multimedia communication protocol used in Integrated Services Digital Network (ISDN).
　　——Specific protocol standards include: H.255.0 (call processing protocol), H.245 (control processing protocol), H.261 and H.263 (video processing protocol), T.12O (data processing protocol). In the IP telephone system, this part of the work mainly completes the following tasks:
　　——. Real-time audio encoding processing
　　——. Control Protocol
　　——. Data Transmission Protocol

　　——2.5—Data exchange between gateways

　　——Data exchange between gateways is a very important and very difficult technology in the development of IP telephone systems. Although IP phone manufacturers all claim that their equipment meets the basic requirements of the H.323 standard protocol, each manufacturer has its own processing method in the specific processing of H.225, H.245 and Q.931. In terms of the comparison of the IP phone's founding manufacturer Vocaltec and the Capitel IP switch system of Beijing Post and Telecommunications Equipment Factory, the two products meet the H.323 specification, but the processing of the G set in the H.323 protocol is completely different. , Because H.323 does not specify the processing method of G set. Vocaltec company uses a three-step encoding method to encapsulate H.323 packets, while the Capitel IP switch system uses a Chinese standard eight-step encoding method to encapsulate H.323 packets. In this way, the two products are used to encapsulate H.323 packets. When communicating with each other, due to the different encapsulation methods of H.323 packets and different interpretations of the received H.323 packets, incompatibility occurs.

　　 serial number request

　　1 The gateway supports G.729A and G.723 multimedia digital signal codec protocols. G.729A is supported first, and secondly, G.723.1 is supported
　　2 The gateway supports DTMF and MF decoding and encoding (when calling out), and finally you can use the IVR system with mourning
　　3 The gateway supports the intercommunication with the gateway of the switching center
　　 4 gateway supports the "Quick Setup Setting" in the H.323 V2 protocol
　　5 The end-to-end information record code can be transferred between the gateway and the gateway
　　 At the 6th gate, the settlement system and the call confirmation from the operator of the settlement system can be used for authentication.
　　7 Call detailed records can be generated in real time and delivered to the settlement center in real time
　　8 Sales meet iNOW! For gateways and gateways of version 2.0, iNOW must be used first! The authoritative organization, successfully completed iNOW! Organization’s certification program
　　9 For calls to the settlement center, through the settlement center and terminal signals, iNOW! Platform provides compatibility
　　10 Provides interoperability between gateways and settlement systems
　　11 Can transmit gateway routing call signaling and terminal routing call signaling
　　12 At least 24 hours, the local gateway clock can be synchronized with an accurate and reliable time source
　　13 Using the following algorithm, the terminal source code can be generated in the settlement system CDR: H.323 1000 Q.931
　　14 The gateway supports the T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27, V.17
　　15 Regarding the settlement system call, the gateway will ensure the integrity of the information
　　——In view of the current situation where there is no universal international standard, in January 1999, Lucent, Itexc and Vocaltec jointly formulated the IP telephony industry standard-iNow! protocol, which mainly includes five aspects:
　　——.Gateway to Gateway intercommunication requirements
　　——.Gatekeeper to Gatekeeper intercommunication requirements
　　——.Gatekeeper to the settlement center intercommunication requirements
　　——. Phone to Phone service requirements
　　——.FAX to FAX service requirements
　　——While meeting the above requirements, the information interaction processing process must be completed under the control of the settlement center. IP telephony operators in different regions can complete various certifications and exchanges through the settlement center. In the iNow! agreement, the connection and disconnection process is also strictly defined, which ensures that products from different manufacturers are compatible with each other in the connection and disconnection process. The iNow! protocol specifies various detailed message formats while specifying algorithms and information exchange specifications. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of IP phone manufacturers can be compatible with each other.
　　——However, good wishes do not equal reality. Since its birth, the INOW! Agreement has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol, and it is still limited to the scope of the H.323 protocol. The incompleteness of the network layer and the lack of guarantee for transmission of the H.323 protocol cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by I TU. Therefore, although the iNOW! protocol has been launched for more than a year, many manufacturers have supported the protocol, but the products of manufacturers claiming to support the protocol are not compatible with each other.
——China's IP phone system, after more than half a year of trial operation, in response to the current problems of the IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and user problems, China's IP phone compatibility has been formulated With the cooperation of relevant units, the network access test and certification of IP telephone equipment has been carried out, and good results have been achieved.

　　3 Technology Convergence

——Network development is evolving towards broadband and intelligence. The current integration of circuit switching and packet switching is the inevitable result of this trend. Due to the high efficiency and low cost of packet switching, it will gradually replace the current Circuit-switched network. Multiple access networks (wireless, x DSL, Cable, optical access, etc.) will become a unified packet-switched backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of circuit switches in the past are constantly being decomposed, and interfaces are being standardized. M GCP (Media Gateway Control ProtocolThe protocol makes the interface between the IP network and the PSTN network have a unified specification, and the IPST (Internet Protocol Standard Transmit) protocol makes the circuit-switched signaling have a unified implementation method in the IN network. This makes the distributed call processing structure in the IP telephony field possible, laying a solid network foundation for the current and future applications of the IP telephony system. The IP telephone system developed at this stage is called the "unified stage" IP telephone system.

　　-The most notable feature of IP telephony in the unified phase is that various IP telephony devices are compatible with each other, and the circuit switching idea is extended to the entire network, so that operators can perform barrier-free switching on the entire IP network. The protocols represented by M GCP and IPST unify H.323 and iNOW! The protocol is standardized, and the interface signaling between the IP network and the PSTN network is standardized (IPST protocol). As we know, IP telephone system is generally divided into three-layer structure, namely: connection layer, control layer and business management layer.

　　——The connection layer is responsible for establishing and realizing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The related protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of the voice signal and establish an appropriate bearer connection. The business management layer mainly completes the operator's business control, such as user management, billing, settlement, and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol.

　　——MGCP is a protocol standard defined for all gateways between PSTN and IP networks. The most typical applications are IP telephone gateways and dial-up access servers. Therefore, the future structure of the IP telephone gateway and the dial-up access server is very similar. The difference is that the IP telephone gateway completes the bundling of PSTN voice channel resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice channel resources and IP sessions. Therefore, the future dial-up access server will be able to automatically identify IP access and IP phone (or fax) access, so as to dynamically realize channel allocation and resource bundling on demand.


　1. The concept of IP telephony

　　IP phone is usually called Internet phone or network phone. As the name implies, it is to make a phone call through the Internet. Broadly speaking, it should be called Internet telecommunications because it includes multiple telecommunications services such as voice, fax, and video transmission.

　　2. Basic principles of IP telephony

　　 The voice of the IP telephone is transmitted by the IP (Internet/Intranet) data network based on router/packet switching. Because the Internet uses the "store-and-forward" method to transfer data packets, it does not monopolize the circuit, and the voice signal is greatly compressed, so the bandwidth occupied by the IP phone is only 8kbit/S-10kbit/S, plus The billing method of packet switching has nothing to do with the distance, which naturally greatly saves long-distance communication costs. The Internet is composed of many different computer networks interconnected all over the world. The emergence and popularization of Internet has greatly changed the way people communicate and communicate. Internet uses standard TCP/IP protocol to realize mutual communication and data exchange between computers.
　　 TCP/IP protocol is responsible for queuing the IP data packets to be transmitted to the network. Each packet contains address and data reorganization information to ensure data security and correct data packet exchange. IP Telephony uses the Internet as the main transmission medium for voice transmission. First, the voice signal is transmitted to the IPTelephony gateway through the public telephone network; then the gateway converts and compresses the voice signal into a digital signal and transmits it to the Internet; and these digital signals are transmitted to the gateway of the other party through a low-cost network all over the world. Then the gateway will restore the digital signal to an analog signal, input it into the local public telephone network, and finally transmit the voice signal to the recipient.

　　3. The key equipment of the IP telephone system—a gateway

The gateways located in various places are represented by a unique IP address, which is a bridge between the two communication transmission methods, and the "switching office" on the Internet to realize the interconnection and communication between remote telephones. On one side, The gateway is connected to a traditional circuit-switched network (Circuit-switched Network) such as the public switched telephone network (PSTN), and can communicate with any external telephone. On the other side, the gateway is connected to a packet-switched network (Packet-Switched Network) such as the Internet, Intranet, etc., can communicate with any computer connected to the network. In the entire Internet Phone system, gateways are distributed all over the world, processing the local PSTN network and Internet access and conversion. The gateway receives standard telephone signals and is digitized After being compressed to a large extent, use the IP protocol to packetize and send it to the Internet, find the transmission route, and send it to the destination through the Internet. On the contrary, receive the data packet transmitted from the Internet and transfer it to the telephone network system. Access and transfer The telephone network system can be carried out at the same time to realize full-duplex (two-way) communication. For example, a long-distance call to San Francisco is made in Beijing, and in Beijing, an ordinary public telephone is connected to the local gateway through the PSTN, and the local gateway performs specific compression on the data Algorithm processing, organized into IP packets containing data such as calling number, called number, time, call information, etc., and analyze the called number, map it to an IP address according to the routing table, and send it to the IP address through routing ( Such as San Francisco) corresponding remote gateway. In the called party San Francisco, the remote gateway receives the IP data packet transmitted by the Beijing local gateway, performs the reverse process of decompression, and then sends it to the local PSTN network. In this way, it is realized Real-time communication between the two places. The communication cost included is only the local ordinary telephone fee in Beijing, the Internet communication fee and the local telephone fee in San Francisco. Because the Internet communication fee is relatively low, the long-distance call fee has been greatly reduced.

　　4. Voice quality of IP phone

　　 Voice quality basically depends on two factors: one is the speed of the Internet communication line; the other is whether the Internet itself is busy. Compared with the voice quality of ordinary telephones, there are two main differences between the voice quality of IP telephones: First, the voice lags behind, and secondly, there is a slight distortion phenomenon. Those who have used IP phones generally believe that the voice quality is better than expected, generally speaking between ordinary phones and mobile phones. In order to improve the voice quality, the most direct method is to expand the Internet access rate and use good Internet access lines.

　　 5. Several key technologies and standards in the IP telephone system (1)

　　* The basic standard of IP telephone The standard of Internet telephone adopts ITU-T H.323 standard. H.323 is one of ITU's multimedia communication protocol series H.32x. H.323 provides the basic standard for transmitting sound, video and data based on IP network (including Internet). It is a framework protocol. The related transmission, control and sound and video compression standards are shown in the table below (the table also includes Contains a series of multimedia protocols in other networks (ISDN, PSTN)). H.323 defines four basic building blocks in network transmission: terminal, gateway, gatekeeper and multipoint controller (MCU). * Network protocol standards Generally speaking, the call establishment and control of Internet phones are mostly established on the basis of TCP, while the transmission of audio streams is established on the basis of UDP. In order to ensure the real-time transmission, IETF has added several important protocols: RSVP (Resource Reservation Protocol): Generally speaking, it is very difficult to reserve enough bandwidth for multimedia transmission on the IP network. For this reason, IEtF defines the Resource Reservation Protocol (RSVP). RSVP allows receivers to apply for a specific amount of bandwidth for data transmission. With RSVP, traditional IP networks without QoS (Quality of Service) guarantees obtain QoS guarantees. To be able to use RSVP, H.323 terminals, gateways, MCUs, etc. must be supported, and routers on the IP network must also be supported. RSVP is defined in RFC2205-RF C2209. RTP/RTCP (Real-Time Protocol/Real-Time Control Protocol): RTP is a protocol defined by IETF to transmit audio and video streams. RTP is built on UDP. In the header of RTP, a time stamp (Time Stamp) is defined. ) To ensure the real-time transmission and synchronization of audio and video. RTCP is a protocol for controlling and monitoring RTP and its Qos. H.323 is based on RTP. See RFC1889 and C1890 for the RTP/RCt protocol.

　　6. ​​Several key technologies and standards in the IP telephone system (2)

　　* Voice coding standard H.323 defines a variety of voice transmission, IETF established AVT (Audio/Video Trans port) working group to conduct voice transmission research. At present, the commonly used voice coding bit stream rates in Internet phones are as follows: G.711 64Kbit/s, G722 48-64kbit/s, G.728 16kbit/s, G.723 and G.723.1 5.3kbit/S or 6.3kbit/ S, G.729 and G.729A8 or 13kbit/s. Not transmitting voice data when both parties are not talking can effectively save bandwidth, but in order to prevent the feeling of intermittent sound during mute compression, it is recommended to add background noise during the mute process. IMTC's VoIP Forum proposes a variable The background noise transfer method of the parameter. *Control module H.323 system control includes: H.245 control, Q.931 call signal control and RAS control. The H.245 control channel is a trusted channel used to carry control information for the operation of H.323 entities. These controls include: performance exchange, opening or closing logical channels, priority requests, flow control information, and basic command instructions. The call signal channel uses Q.931 to establish a connection between two terminals. The RAS signal channel completes registration, access rights, bandwidth changes and status updates. The RAS signal channel is generally established between the terminal and the gatekeeper. If the gatekeeper does not exist, then there is no RAS channel.