CN100452784C - Phonetical gate communication device and method - Google Patents

Abstract

The present invention discloses speech gateway communication equipment. The equipment not only comprises components which the general speech gateway communication equipment comprises, but also comprises a time division digital switching network chip, wherein all the telephone interfaces, relay interfaces and DSP are connected with the time division digital switching network chip by coding-decoding modules (Codec). The present invention also discloses a speech gateway communication method which is characterized in that the method comprises a step of transferring speech signals by an IP (Internet Protocol) network as well as a step of transferring the speech signals by a programmed control switching mode. The method and the equipment of the present invention combine the advantages of the traditional program control switching equipment and the method thereof and the speech gateway equipment and the method thereof to realize the perfection of functions of communication equipment, to effectively reduce the cost and obviously improve the speech quality of internal calls.

Description

A voice gateway communication device and method

technical field

The invention relates to a voice gateway communication device and a voice gateway communication method, in particular to a voice gateway communication device and a voice gateway communication method combining traditional telephone exchange technology and voice transmission technology based on Internet protocol (VOIP technology).

Background technique

Traditionally, for the purpose of communication, almost every enterprise user or group user generally uses a private program-controlled branch exchange (PBX) to achieve intercommunication within a fixed office, and realize communication with external users by accessing an external public network. The working principle of the traditional digital program-controlled switchboard is that firstly, the analog signal of the calling terminal phone is converted into a digital signal in the codec module according to a unified standard; and then encoded to form a data stream of 64K bits per second; The stream enters the time-division digital switching network chip, and the time-division digital switching network can switch this data stream to the interface of the called end phone; then decode the digital signal in the codec module; convert it into an analog signal; and finally transmit it to on the receiving end telephone so that the voice of the other party can be heard. Among them, the time-division digital switching network chip is the core of the program-controlled switch, which is controlled by the central processing unit CPU. The central processing unit CPU includes a set of relatively complex special software system for program-controlled exchange control. The main advantage of using a program-controlled switch is that, for calls between internal users or group users and external users, it can enable more users to fully share less trunk resources to achieve simultaneous calls; As far as the calls between users are concerned, it can make the calls between these internal users not limited by the trunk line resources, but free calls, thus significantly saving costs; in addition, it can also expand many supplementary service functions.

Another device that realizes telephone communication is a device that uses an IP network to transmit voice——VOIP voice gateway device. With the increasing popularization of computers and the wide application of IP technology, many large, medium and small enterprises and many government departments in all walks of life have their own IP networks through leased or self-built methods, and VOIP voice gateway equipment has also emerged as the times require. born. Its working principle is: After receiving the calling voice signal, use the compression algorithm to compress the voice data, then pack the voice data according to the TCP/IP standard, send the data package to the called place through the IP network, and then send the voice data The data packets are stringed together, and after decompression processing, the original voice signal is restored and sent to the called end to achieve the purpose of real-time voice transmission by the Internet. The advantage of using the VOIP voice gateway system is that it can directly use the Internet, has good versatility, and greatly expands the scope of transmitting voice information; saves the cost of making long-distance calls; it can also expand many other functions, such as simultaneous image transmission ,etc.

But above-mentioned two kinds of telephone communication equipments all have deficiencies respectively, namely: on the one hand, the shortcoming of traditional PBX is that the PBX products that different manufacturers provide contain the proprietary technology of each manufacturer, poor compatibility between products, poor expansibility; , the traditional PBX cannot simply transmit voice through the IP (Internet Protocol) network, that is, it does not have the functions that VOIP equipment has. The VOIP technology is a revolutionary technology that realizes the transition from a traditional closed telecommunication system to an open, IP-based telecommunication platform. It opens up the traditional exchange control platform systems in the past, making new products and new applications emerge in an endless stream. Very broad. If the traditional PBX is not combined with IP technology, its application will be greatly limited. In addition, the cost of PBX equipment is high, and the cost of using PBX to make long-distance calls is also high. On the other hand, voice gateway equipment does not have many functions of digital program-controlled exchanges, especially many functions of private telephone exchanges, which leads to the fact that in many applications, the functions provided by voice gateways cannot meet the needs of users. For example, many supplementary services are stipulated in national standards on telephone communication, and the voice gateway cannot realize all supplementary service functions, which will also limit its application range. At the same time, when a telephone user of the voice gateway calls another telephone user inside the voice gateway, the call quality deteriorates. Because whether it is a call inside or outside the gateway, the voice signal must be transmitted after the compression process, which will not only inevitably cause distortion, resulting in a decline in call quality, but also the voice gateway’s compression and decompression of the voice signal. A large amount of processing capacity of the system, especially the DSP processing capacity of the system, will reduce the processing capacity of the system. Ask for more. In addition, in order to use the functions provided by the two at the same time, users have to purchase a set of PBX equipment and VOIP equipment at the same time, which increases the cost of fixed assets and maintenance costs.

Contents of the invention

The purpose of the present invention is to provide a novel voice gateway communication device and a novel voice gateway communication method combining traditional PBX technology and VOIP technology.

According to an object of the present invention, a kind of novel voice gateway communication equipment combining traditional PBX technology and VOIP technology is provided, including several telephone interfaces and relay interfaces, digital signal processor (DSP), central processing unit ( CPU) and Ethernet interface; one end of the Ethernet is connected to the central processing unit (CPU), and the other end is connected to the external data network; it is characterized in that it also includes a time-division digital switching network chip, all telephone interfaces and relay interfaces Both are connected to the time-division digital switching network chip through the codec module; the DSP is also connected to the time-division digital switching network chip. In this way, all telephone interfaces, relay interfaces, and DSP interfaces can exchange signals with each other through the time-division digital switching network in the time-division digital switching network chip. When the called object is located outside the gateway, under the control of the central processing unit CPU, the telephone voice is converted into a digital signal through the codec module, and then transmitted to the DSP through the time-division digital switching network, and the DSP compresses the digital signal to form a voice Afterwards, the central processing unit CPU sends this voice packet to the external data network, and arrives at the called object by the external data network. Therefore, the present invention has the function of transmitting voice over the Internet (VOIP). And when the called object is located inside the gateway or the trunk interface, under the control of the central processing unit CPU, the voice signal is converted into a digital signal, and after encoding, it is directly switched to the corresponding codec of the called end through the time-division digital switching network , decoded and restored to an analog voice signal, and finally reaches the called object. Therefore, the present invention has all the functions of the digital program-controlled exchange.

According to another object of the present invention, a kind of novel voice gateway communication method combining traditional FBX technology and VOIP technology is provided, as shown in Figure 5, comprises the following steps:

a. Receive the call signal from the telephone interface or trunk interface, convert the analog voice call analog signal into a digital signal and encode it through the codec module;

It is characterized in that it also includes the following steps:

b. Judging whether the called object is located inside or outside the gateway; if the called object is located outside the gateway, then perform the following step c; if the called object is located inside the gateway or a trunk interface, then perform the following step d;

c. It is transmitted to the digital signal processor DSP through the time-division digital switching network, and the DSP compresses the digital signal to form a voice packet, and then the central processing unit CPU sends the voice packet to the external data network, and reaches the called party through the external data network object;

d. Under the control of the central processing unit CPU, the coded digital signal is directly switched through the time-division digital switching network to the codec module corresponding to the called object, restored to an analog voice signal, and finally reaches the called object.

The invention adds a time-division digital switching network to the common voice gateway, and realizes the communication function through the software control process flow of the central processor. Compared with the traditional PBX, the present invention has obvious advantages, such as: the telephone network and the computer network are unified into a whole, and one machine is dual-purpose or multi-purpose. In addition to providing services for traditional telephone users, it can also conveniently serve Internet users provide services; at the same time, in addition to providing telephone program-controlled switching functions, it can also provide voice gateway communication functions; it has significantly saved equipment investment and operating costs for users; the new hardware platform and highly integrated system functions can be more convenient and easy. Realize value-added services, such as establishing a call center, implementing VOIP, etc.; move the dedicated communication platform to a computer platform that is generally familiar to the public, with open standards, stronger interoperability, versatility and practicability; use, configuration and maintenance Simpler; more powerful and highly integrated, a single system can complete the functions that need to be equipped with many devices outside the traditional PBX, such as automatic attendant desk, voice mail, etc.; due to the use of computer platforms, the expansion of the system and Upgrade is easier and saves investment; application development is convenient and simple. And compare with general VOIP voice gateway product, the present invention can realize voice exchange function; Owing to exempted from data compression and decompression processing during internal conversation, therefore reduced the distortion of voice, improved conversation quality, reduced DSP simultaneously. The occupation of the channel improves the processing capacity of the DSP system and effectively reduces the cost of the system; in addition, it can realize the telephone value-added services that general VOIP products do not have, such as call hold, call waiting, three-party negotiation and other supplementary services; in the same In the case of the number of users, the cost performance of the present invention is high. Therefore, the present invention can not only overcome the respective weak points of PBX equipment and VOIP equipment, but also combine the advantages of both PBX and VOIP, that is, while realizing the VOIP function, the functions of all program-controlled exchanges are provided again, which can satisfy Various needs of users in different occasions. Therefore, the present invention not only realizes the perfection of the function of the communication equipment, effectively reduces the cost, but also obviously improves the voice quality of the internal communication.

Description of drawings

Fig. 1 is the line connection block diagram of the special program-controlled exchange of prior art.

Fig. 2 is a block diagram of line connections of a VOIP voice gateway device in the prior art.

Fig. 3 is a block diagram of line connection of the voice gateway communication device of the present invention.

Fig. 4 is a diagram of functional modules of a time-division digital switching network chip and a central processing unit (CPU) of the device of the present invention.

Fig. 5 is a flowchart of the method of the present invention.

FIG. 6 is a flowchart of a scheme of the voice gateway communication method of the present invention.

Detailed ways

The equipment of the present invention will be further described in detail below in conjunction with the accompanying drawings.

Fig. 3 is the line connection block diagram of equipment of the present invention, and voice gateway communication equipment of the present invention comprises several telephone interfaces and relay interface, digital signal processor (DSP), central processing unit (CPU) and ethernet interface; Ethernet one end It is connected to the central processing unit (CPU), and the other end is connected to the external data network; it also includes a time-division digital switching network chip, and all telephone interfaces and trunk interfaces are connected to the time-division digital switching network chip through a codec module; DSP is also connected with this time division digital switching network chip. In this way, all telephone interfaces, relay interfaces, and DSP interfaces can exchange signals with each other through the time-division digital switching network in the time-division digital switching network chip. Among them, the telephone interface, trunk interface, digital signal processor (DSP), Ethernet interface and codec module are all common components in the field of PBX technology and VOIP technology, and can be purchased commercially. Time-division digital switching network chip and central processing unit (CPU) are then installed with logic circuit and control software system designed and manufactured by the function requirements of the equipment of the present invention, and their modules are shown in Fig. 4 . On the basis of understanding the content of the present invention, those skilled in the art can easily know the specific content of each module and know how to implement them.

Fig. 6 is a flowchart of a solution of the voice gateway communication method of the present invention, including steps:

Step 1: The phone goes off-hook, and the CPU controls the time-division digital switching network to send a dial tone to the phone interface;

Step 2: The user dials, the CPU controls the time-division digital switching network to cut off the dial tone, and the database software performs number analysis. If it is a VOIP call, then perform the following steps 3 and 4, otherwise perform steps 5 and 6;

Step 3: The VOIP task sends the call to the network through the VOIP protocol, reaches the called gateway, and the telephone interface is connected to the DSP through the time-division digital switching network;

Step 4: The called party answers, the caller and the called party talk, DSP compresses the voice, the CPU sends the voice to the called party through the network interface, and then perform the following step 7

Step 5: If it is an internal call, the called party will ring; if it is a trunk, send the call information to the other party. Connect the calling phone to the ringback tone through the time-division digital switching network;

Step 6: The called party answers, connects the calling party and the called party through the time-division digital switching network, and communicates between the calling party and the called party.

Step 7: The calling party hangs up, cuts off the switching network, sends a call release signal to the called party, and releases system resources.

For realizing above steps, the control system that the inventive method uses adopts real-time operating system and realizes with multitasking, the task module that creates in the system comprises network task, interface communication task, call processing task, database task, VOIP protocol task, DSP control communication In addition, tasks may also include accounting tasks, maintenance configuration tasks, and the like. Among them, the specific content of each task is:

1. Network task: responsible for receiving and sending data from the network to realize network communication.

2. Interface communication task: complete the communication between the interface board and the system software, collect hardware events, and convey the control of the system software to the interface board.

3. Call processing tasks: complete the control of telephones and trunks, realize the communication protocol and supplementary services of digital program-controlled switches, etc. The limited message status level mechanism is adopted to complete the control of the telephone interface and trunk interface. The telephone interface, trunk interface, switching network, and DSP channel transmit the voice signal flow to each other under the control of the call processing task, and complete the function of the digital program-controlled exchange. It can also work closely with the VOIP protocol to realize the VOIP function.

4. Database tasks: Serving call processing tasks and VOIP protocol tasks, providing functions such as number analysis, routing selection, resource management, and resource allocation.

5. The VOIP protocol task realizes the function of VOIP, realizes the communication with other voice gateways, cooperates closely with the call processing task, and provides fax service while providing voice communication.

6. Billing tasks collect all billing raw information from call processing tasks: and VOIP protocol tasks, and send them to the billing server, so that all calls can be billed on the billing server, suitable for operators or virtual operators operating requirements.

7. Maintenance and configuration tasks: complete system data maintenance and management, and provide various maintenance methods, including HTTP, Telnet, SNMP and other maintenance methods. The maintenance configuration task should cooperate closely with the database task to complete the configuration and maintenance of the system together.

8. DSP control communication task: complete the control of the digital signal processor (DSP), and realize the communication between the system software and the DSP, can deliver the compressed voice packet output by the DSP to the network task, and can also receive the network task The voice data packets are sent to DSP, decompressed by DSP and output to digital switching network, and output to each interface (telephone interface and trunk interface) by digital switching network to realize VOIP function.

Claims (3)

1. a voice gateways communication apparatus comprises several telephony interfaces and trunk interface, digital signal processor, central processing unit and Ethernet interface; Ethernet one terminates on the central processing unit, and the other end is received on the outer data network; It is characterized in that, also comprise a time-division digital switching network chip, all telephony interfaces all are connected with time-division digital switching network chip by coding/decoding module with trunk interface, digital signal processor also therewith time-division digital switching network chip be connected, under the control of central processing unit, when being positioned at the gateway outside by call object, after call voice converts digital signal to by coding/decoding module, be transferred to digital signal processor by the time-division digital switching network and compress processing, send to by call object by outer data network by central processing unit then; When being positioned at intra-gateway or trunk interface by call object, voice signal converts the coding/decoding module that directly exchanges to the called side correspondence after the digital signal by the time-division digital switching network to, and decoding reduction back arrives by call object.

2. voice gateways communication apparatus according to claim 1, it is characterized in that: under the control of central processing unit, all telephony interfaces, trunk interface, digital signal processor interface can both be carried out intercoursing of signal by the time-division digital switching network in the time-division digital switching network chip.，

3. voice gateways means of communication comprise:

A. be subjected to call signal from telephony interface or trunk interface termination, the analog voice call analog signal conversion is become digital signal and encode by coding/decoding module;

It is characterized in that, further comprising the steps of:

B. judge by call object and be positioned at still outside of intra-gateway; If be positioned at the gateway outside, then carry out following steps c by call object; If be positioned at intra-gateway or trunk interface, then carry out following steps d by call object;

C. be transferred to digital signal processor by the time-division digital switching network, compress processing by digital signal processor word signal and form voice packet, by central processing unit this voice packet is sent to outer data network then, arrive by call object by outer data network;

D. under the control of central processing unit, the digital signal of having encoded is directly arrived by the coding/decoding module of call object correspondence by time-division digital switching network exchange, be reduced into analog voice signal, finally arrive by call object.

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