CN100382514C - Voice quality testing method for network voice device - Google Patents

Abstract

The present invention provides a method for testing a network voice device. The method does not require the purchase of expensive special test equipment, nor is it limited to being implemented near the network voice device being tested. The network voice device being tested does not require special hardware arrangements, only simple wiring is required. The method provided by the present invention can also be used to test interface quality and line quality at the same time without preparing two different sets of test equipment and environments. The method provided by the present invention can be performed through a local area network and a wide area network (such as the public Internet), so the network voice device being tested can be located at any location that can be reached through the public Internet.

Description

Voice Quality Testing Method for Internet Voice Devices

technical field

The invention relates to detecting network voice devices, in particular to a method for detecting voice quality of network voice devices.

Background technique

With the popularity and growth of the Internet, various brand-new network applications continue to appear, which also have many new changes and impacts on people's lives. Among these new network applications, the most eye-catching ones are various devices, systems, or software called VoIP, Voice over IP (VoIP), and IP telephony. Hereinafter, for the sake of uniformity, these technologies or practices that enable both parties to talk and talk through the Internet are collectively referred to as VoIP.

The reason why VoIP gradually replaces traditional calls via the public switched telephone network (PSTN) is mainly due to economic factors. Traditional PSTN calls are generally billed based on the location of the callee and the length of the call. VoIP has broken the boundaries of distance, country, and region because of the Internet, and network service providers do not target users who belong to it. The data traffic of VoIP is charged, so the cost of VoIP is very low. However, in the past, due to limited network bandwidth, undeveloped related technologies, and different protocol standards, Internet voice was only a game among some players, and it could not enter the home, office and other environments to become a reliable communication medium in people's lives. .

In recent years, these factors that limit the growth of VoIP have been solved one by one, and the application of VoIP has also begun to show explosive growth. But in any case, traditional telephones have existed for more than a hundred years. Traditional telephones are already a part of people’s lives just like water and electricity. There are at least one traditional telephone in a general household, and there are already installed interiors. Internet, and devices like tape recorders and fax machines. Therefore, the fastest growing device on the market is the VoIP device with one end connected to the Internet and the other end connected to the traditional telephone. As shown in Figure 1a, this type of VoIP device 100 has a network port 110 (usually an RJ-45 jack) to pass through a local area network (not numbered), and devices such as ADSL modem 120, Cable modem, etc. It is connected to the public Internet 800 . In addition, the VoIP device 100 also has at least one phone port (usually an RJ-11 jack) 130 to connect to a general phone 140 . Like using a traditional telephone, the user can pick up the handset, dial a number through an interface (not shown) of the device or the number keys of the phone, and then wait for the other party to answer. The other party answers through a device 150 that is the same as or compatible with the VoIP device 100 . The VoIP device 100 can also have another line port 160 connected to the public switched telephone network 900, so as to send a call to the receiver using the traditional telephone network like a traditional telephone. People's speech is a continuous analog signal. Internet voice digitizes, compresses, and encodes this continuous analog signal, and then transmits it sequentially in separate data packets. On the contrary, these data packets are collected sequentially at the receiving end, decoded, and then converted into a continuous analog signal audible to the human ear. During these conversion processes, coupled with various transmission conditions such as delays or losses experienced by packets transmitted through the network, it will affect the call quality experienced by both parties in the call. Traditionally, the quality of these VoIP devices can be mainly divided into two parts: interface quality and line quality. The former is mainly related to the quality of the electrical signal of the voice generated by the VoIP device, including insertion loss (insertion loss), total harmonic distortion (total harmonic distortion, THD), intermodulation distortion (inter-modulation distortion, IMD) , noise value, signal-to-noise ratio (S/N ratio), etc. The latter is related to the VoIP device's ability to withstand and compensate for delays in packet delivery, jitter between packets, damage and loss of packets.

Traditionally, it is necessary to purchase expensive and exclusive equipment to test the quality of these two aspects of the VoIP device, and it is also necessary to set some special arrangements for the VoIP device to be tested. Taking the VoIP device 100 shown in Figure 1a as an example, if the quality of the interface is to be tested, as shown in Figure 1b, a dedicated test device 170 is directly connected to the phone and line ports 130, 160 of the VoIP device 100 using RJ- 11 cables (not numbered) are connected. It is also necessary to arrange a loop back (loop back) between the telephone port and the line port 130, 160 inside the VoIP device 100 (shown by a dotted line in the figure). When performing a test, the test equipment 170 sends a special test signal through the line port 160 into the VoIP device 100 , and the test signal is sent back to the test equipment 170 through the return connection and the telephone port 130 . The test equipment 170 can calculate the relevant interface quality parameters by comparing the test signals sent initially and received later. In addition to the investment of special test equipment and the setting of the circuit of the device under test, this test method has other disadvantages, such as the test equipment and the device under test must be located at the same place and not too far apart, in order to connect the two are connected. If the device under test is located at another location, then the test equipment will need to be moved there for testing.

If it is to test the line quality, as shown in FIG. 1c, it is more complicated, and additional special test equipment 180, 190 is required to be connected to the telephone port 130 of the VoIP devices 100 and 150 respectively. Special test equipment 180 sends out some known test signals, via VoIP device 100, the public Internet or a simulated network 810 (can provide different degrees of delay, packet loss rate, etc.), VoIP device 150 to the test device 190. Then by comparing known test signals, relevant quality data can be obtained. In addition to the investment in special test equipment and the setup of a simulated network, this test method also has other disadvantages. For example, the test equipment and the device under test must be located at the same place and not too far apart to connect the two.

Contents of the invention

In order to overcome the shortcoming of known testing VoIP device, the present invention proposes a kind of method of testing VoIP device, this method does not need to buy expensive special test equipment, is not limited to must implement near the VoIP device under test, The VoIP device under test does not require any special hardware arrangements, only a simple connection is required. The method proposed by the present invention can also be used to test the quality of the interface and the quality of the line at the same time, without the need to prepare two sets of different test equipment and environment.

The test method proposed by the present invention can be executed on a general computing device in the form of a software module. The computing device is connected to the VoIP device under test through a standard local area network, and then the test is performed through the execution of the software. As long as one or more VoIP devices to be tested are connected to this local area network, they can be tested one by one by the method proposed by the present invention, so the test environment and settings are very simple, and it is not necessary to test the network voice devices under test. What modification does the speech device implement.

Except local area network, the method that the present invention proposes also can be carried out by wide area network (like public Internet), so the VoIP device tested can be positioned at any place that can be reached by public Internet, so the present invention can be used to test VoIP devices installed in remote customer environments without the need to send personnel carrying devices to customer locations for testing.

The method proposed by the present invention can be applied to the VoIP device with one or more telephone ports, and can also be applied to the VoIP device without or with a line port. Taking the most common VoIP device with one phone port and one line port as an example, as long as the two ports are connected in series with a standard RJ-11 cable, the related tests of interface quality and line quality can be carried out. There is no need to set up two different environments.

The above and other purposes and advantages of the present invention will be described in detail below in conjunction with the accompanying diagrams, detailed descriptions of the embodiments and the protection scope of the patent application. However, the attached icons are marked purely for illustrating the present invention, and should not be regarded as limiting the protection of the present invention. For the protection scope of the present invention, please refer to the patent scope in the appended claims.

Description of drawings

FIG. 1a is a schematic diagram of connection of a known VoIP device.

FIG. 1b is a schematic diagram of connection of a known voice over network device testing interface quality.

FIG. 1c is a connection schematic diagram of a known VoIP device testing line quality.

FIG. 2a is a schematic diagram of a local area network according to the first embodiment of the present invention.

FIG. 2b is a schematic diagram of a second embodiment of the present invention based on a local area network.

FIG. 2c is a schematic diagram of a third embodiment of the present invention based on a local area network.

FIG. 2d is a schematic diagram of a fourth embodiment of the present invention based on a wide area network.

FIG. 3 is a schematic diagram of a coupling device according to a fifth embodiment of the present invention.

in the picture

100 VoIP device 110 Network port

120 ADSL modem 130 phone port

140 Telephone 150 VoIP device

160 Line ports 170 Test equipment

180 Test Equipment                     190 Test Equipment

200 VoIP device 202 Network port

210 Test procedure 212 Use interface

214 Test data generation module 216 Data analysis module

220 Wired Ethernet Local Area Network 230 VoIP Device

232 Line port 234 Phone port

236 Network port 238 Encoding and decoding module

240 RJ11 cable 250 VoIP device

300 Coupling device 310 VoIP device

312 phone port 314 phone port

322 Tip wire                                                                                                                        

332 Tip Wire 334 Ring Wire

800 Public Internet Network 810 Analog Network

900 Public Switched Telephone Network

Detailed ways

The method for testing the VoIP device proposed by the present invention can be installed on a general computing device in the form of a software module for execution. Such computing devices include (but are not limited to) desktop computers, notebook computers, and even PDAs. The method proposed by the present invention can also be implemented in a dedicated independent device in a solid state. The feature of these independent or computing devices is that they need to have a network interface, which can communicate with the VoIP device to be tested through a network. Usually the network interface is an RJ-45 network port, and then the computing device and the VoIP device to be tested are connected through a wired Ethernet network. In other embodiments, the network interface may also be an optical fiber interface, or a wireless network supporting the 802.11 protocol. That is to say, the present invention does not limit the specific method and network used for connecting the voice device under test, as long as various network interfaces and networks can be used to connect the computing device and the voice device under test.

FIG. 2a is a schematic diagram of a local area network according to the first embodiment of the present invention. As shown in FIG. 2 a , the method proposed by the present invention is installed in the computing device 200 in the form of a test program 210 . The test program 210 mainly includes a user interface 212 , a test data generation module 214 , and a data analysis module 216 . After the test program 210 is executed, the user interface 212 allows the user to input relevant test parameters, activate test execution, and display test results through the keyboard and screen (not shown) of the computing device 200 . Many details of the computing device 200 are omitted, such as processor, memory, bus, hard disk, operating system, etc., because these details should be well known to those skilled in the art, so they are not repeated here.

The computing device 200 is connected with a network port 202 and a wired Ethernet LAN 220 . The test data generation module 214 generates a series of specific test data packets (not shown), and transmits them to the VoIP device 230 under test via the network port 202 and the local area network 220, and the packets processed by the VoIP device 230 will pass through the local area network. 220 and the network port 202 are received by the data analysis module 216 and compared with a series of test packets previously sent out to analyze the quality of the VoIP device 230 . The local area network 220 also has considerable details omitted, such as switches forming part of the local area network 220 .

The VoIP device 230 under test has a line port 232 and a phone port 234 . This is the most common small VoIP device. The line port 232 generally has the so-called FXO interface, that is, the interface of the general telephone set to the central office, and has functions for receiving ringing (ring) signals, hanging up (on-hook), answering (off-hook), and sending out DTMF key sounds. The function is an interface used to connect with the public telephone switching network. The phone port 234 generally has a so-called FXS interface, that is, the interface of the central office dialogue phone, which has the functions of providing loop current, sending out vibration signals, providing dial tone, and detecting hangup and answering. , is the interface used to connect with the phone.

Please note here that the line port 232 of the general VoIP device 230 has two functions, one is to allow a phone connected to the phone port 234 to directly send a call to the public switched telephone network connected to the line port 232, and the other is to allow a call from the network An incoming VoIP call at port 236 is forwarded to the public switched telephone network connected to line port 232 . For the latter, the earliest packet of the VoIP incoming call will record the telephone number of the recipient to be dialed to the public switched telephone network. After the VoIP device 230 receives this packet, it will pass through the FXO interface of the line port 232 , simulating a general telephone to pick up the handset (off-hook), detect the dial tone, send out a key tone representing the phone number of the called party, and then connect the called party and the calling party through the encoding and decoding module 238 . On the other hand, another way to send a call to the same or compatible device on the Internet via the phone port 234 is that when the handset of the phone connected to the phone port 234 is picked up (off-hook), the FXS interface of the phone port 234 detects After this situation, a dial tone is provided to the phone connected to the phone port 234. At this time, the caller can directly use the keys of the phone to input the network address of the same or compatible device of the other party, and the VoIP device 230 will send the calling phone through the phone port. 234. The encoding and decoding module 238 is connected with the other party's device.

As shown in FIG. 2 a , the line port 232 and the phone port 234 are connected by an RJ11 cable 240 . Such a simple connection plus the characteristics of the FXS and FXO interfaces completes a simple loop back connection. Under such back connection, for the line port 232, the telephone port 234 plays the role of the public telephone switching network, and for the telephone port 234, the line port 232 plays the role of the telephone.

The operation mode of the test method proposed by the present invention is described as follows. Firstly, the test packet generated by the test data generation module 214 indicates a telephone number to be sent to the public switched telephone network through the line port 232 , and this telephone number is actually the network address of the computing device 200 . This test packet is then sent to the VoIP device 230 . After VoIP device 230 receives this packet, according to the flow of the aforementioned FXO interface, it first picks up the handset (off-hook, off-hook) by simulating a normal telephone, and detects off-hook via the line 240 connected back, and the phone port 234, so Send a dial tone, after the line port 232 detects the dial tone, the telephone number of the network address of the computing device 200 is sent out with a DTMF key tone, after the phone port 234 receives the network address, it tries to communicate with the network address where the network address is located. The device 200 establishes a channel for voice over Internet calls. So far, the network computing device 200 and the network voice device constitute a test loop as shown by the dotted line in FIG. 2a.

Next, the test data generation module 214 continues to generate and send specific test data packets. These packets are first decoded into analog signals by the encoding and decoding module 238 through the path shown by the dotted line, and then pass through the line port 232 and the telephone port 234. , and then encoded by the encoding and decoding module 238 into a data packet, which is received by the data analysis module 216 through the local area network 220 . These packets are then compared and analyzed with the original test packets, and the quality of the interface and line can be analyzed. Please note that during this test, the network addresses of the VoIP device 230 and the computing device 200 must be known. Moreover, the interface functions of the above-mentioned FXO and FXS are all possessed by the general VoIP device 230 , and no additional setting and modification of software and hardware are required. In addition, in terms of data analysis, any known technology or algorithm can be used. Since the content of the algorithm is not the spirit of the present invention, it will not be described in detail here.

From the above description, it can be clearly seen that the present invention can detect the interface quality and line quality of the VoIP device by utilizing the interface and functions of the VoIP device itself and a very simple connection method. For other VoIP devices to be tested, just know their network addresses, connect in this way, and execute the test program again. There is no need to purchase special test equipment, and the setup of the test environment is very simple.

When the VoIP device 230 provides more than one phone port 234, the connection method shown in FIG. 2b can be used to test each phone port 234 one by one. When the VoIP device has only one phone port 234, the connection mode shown in FIG. 2c can be used, and another VoIP device 250 with an FXO interface can be used for testing.

Based on the same architecture, the present invention can also use the public Internet 800 to replace the local area network 220, as shown in FIG. 2d. Under such a framework, the present invention can detect remote VoIP devices through the public Internet.

The above embodiments all need to utilize a FXO interface (whether it is directly located in the VoIP device under test or utilizes the FXO interface of another VoIP device), for VoIP devices without FXO interfaces, or cannot When obtaining another VoIP device with FXO interface, the present invention can test the FXS interface of the VoIP device through a coupling device as shown in FIG. 3 .

As shown in FIG. 3 , the coupling device 300 is a simple circuit composed of passive components such as inductors and capacitors (not labeled). In the FXS interface of the phone port 312 of the VoIP device 310 under test, two wires 322, 324 known as Tip and Ring, and the Tip and Ring wires 332, 334 in the FXS interface of the phone port 314, as shown in Figure 3 Connecting to coupling device 300 in the manner shown results in loop current flowing through wires 322 and 324 at phone port 312 and loop current flowing through wires 332 and 334 at phone port 314 . In other words, after connecting the coupling device 300, both the phone ports 312 and 314 will be on-hook, and the inductance in the coupling device 300 will enable the electrical signal of the voice on the wires on both sides to be coupled from one side to the other side, so that Phone ports 312 and 314 allow direct conversation. Then the computing device (not shown in the figure) can send packets to the phone ports 312 and 314 respectively, and first requires both phone ports to stop sending dial tone (in some embodiments, if the dial tone can be filtered, this step can be omitted ). Please note that there is a field in the packet to indicate which phone port it is for, and the test program (not shown) also needs to be told which two phone ports are currently being tested. Then the test data generation module of the test program can send test data packets for any one of the phone ports. Take the phone port 312 as an example. The coupling device 300 and the telephone port 314 are encoded into data packets by the encoding and decoding module, and then received by the data analysis module of the test program through the network. These packets are then compared and analyzed with the original test packets, and the quality of the interface and line can be analyzed. In addition to being applicable to VoIP devices with multiple phone ports, the present invention can connect the phone ports of two VoIP devices in the manner shown in FIG. 3 for testing. The same test methods shown in FIG. 3 are also applicable to the environment of a local area network or a wide area network.

Through the detailed description of the preferred embodiments above, it is hoped that the characteristics and spirit of the present invention can be described more clearly, and the scope of the application is not limited by the preferred embodiments disclosed above. On the contrary, the purpose is to cover various changes and equivalent arrangements, all of which belong to the scope of protection of this application.

Claims (12)

1. method for testing speech quality of network speech apparatus, be imposed in an arithmetic unit, and the speech quality by a network measuring one network speech apparatus, this arithmetic unit have a network port with this network on-line, this network speech apparatus comprises a network port at least and has the phone port of FXS interface with the line port and that has a FXO interface with this network on-line,, and this method of testing comprises the following step at least:

This network port of this arithmetic unit and this network on-line have caused one first network address, this network port of this network speech apparatus and this network on-line have caused one second network address, and with this line port and this phone port of online this network speech apparatus of a cable;

This arithmetic unit produces a package and is sent to this second network address, this package records this first network address, and after this network speech apparatus that is positioned at this second network address is received this package, cause this network speech apparatus via its line port its phone port to be carried out a dial action, the number that this dial action is dialled is this first network address;

This arithmetic unit is accepted by the voice-over-net calling on-line that this network speech apparatus sent that is positioned at this second network address;

This arithmetic unit produces the good a plurality of test packets of decision in advance, and is sent to this second network address; And

Receive and analyze the plurality of packets of passing back by this network speech apparatus that is positioned at this second network address.

2. method for testing speech quality of network speech apparatus as claimed in claim 1, wherein, this network is a LAN.

3. method for testing speech quality of network speech apparatus as claimed in claim 1, wherein, this network is a Wide Area Network.

4. method for testing speech quality of network speech apparatus, be imposed in an arithmetic unit, and the speech quality by a network measuring one first network speech apparatus, this arithmetic unit have a network port with this network on-line, this first network speech apparatus comprises a network port at least to have the phone port of FXS interface with this network on-line and, and this method of testing comprises the following step at least:

Prepare one second network speech apparatus, this second network speech apparatus comprises a network port at least to have the line port of FXO interface with this network on-line and;

This network port of this arithmetic unit and this network on-line have caused one first network address, this network port of this first network speech apparatus and this network on-line have caused one second network address, this network port of this second network speech apparatus and this network on-line have caused one the 3rd network address, and connect this line port of this second network speech apparatus and this phone port of this first network speech apparatus with a cable;

This arithmetic unit produces a package and is sent to the 3rd network address, this package records this first network address, and after this second network speech apparatus that is positioned at the 3rd network address is received this package, cause this second network speech apparatus via its line port this phone port of this first network speech apparatus to be carried out a dial action, the number that this dial action is dialled is this first network address;

The voice-over-net calling on-line that this arithmetic unit acceptance is sent by this first network speech apparatus that is positioned at this second network address;

This arithmetic unit produces the good a plurality of test packets of decision in advance, and is sent to the 3rd network address; And

Receive and analyze the plurality of packets of passing back by this first network speech apparatus that is positioned at this second network address.

5. method for testing speech quality of network speech apparatus as claimed in claim 4, wherein, this network is a LAN.

6. method for testing speech quality of network speech apparatus as claimed in claim 4, wherein, this network is a Wide Area Network.

7. method for testing speech quality of network speech apparatus, be imposed in an arithmetic unit, and the speech quality by a network measuring one network speech apparatus, this arithmetic unit have a network port with this network on-line, this network speech apparatus comprises a network port at least to have the phone port of a plurality of FXS interfaces with this network on-line and, and this method of testing comprises the following step at least:

This network port of this arithmetic unit and this network on-line have caused one first network address, this network port of this network speech apparatus and this network on-line have caused one second network address, one first phone port of this network speech apparatus and one second phone port, the electrical couplings that is connected with each other is also carried machine;

This arithmetic unit produces the good a plurality of test packets of decision in advance, and is sent to this second network address, and this plurality of packets records the address of this first phone port; And

Receive and analyze the plurality of packets of being passed back by the second phone port of this network speech apparatus that is positioned at this second network address.

8. method for testing speech quality of network speech apparatus as claimed in claim 7, wherein, this network is a LAN.

9. method for testing speech quality of network speech apparatus as claimed in claim 7, wherein, this network is a Wide Area Network.

10. method for testing speech quality of network speech apparatus, be imposed in an arithmetic unit, and the speech quality by a network measuring one first network speech apparatus, this arithmetic unit have a network port with this network on-line, this first network speech apparatus comprises a network port at least to have the phone port of FXS interface with this network on-line and, and this method of testing comprises the following step at least:

Prepare one second network speech apparatus, this second network speech apparatus comprises a network port at least to have the line port of FXS interface with this network on-line and;

This network port of this arithmetic unit and this network on-line have caused one first network address, this network port of this first network speech apparatus and this network on-line have caused one second network address, this network port of this second network speech apparatus and this network on-line have caused one the 3rd network address, this phone port of this first network speech apparatus and this line port of this second network speech apparatus are connected with each other and reach electrical couplings and carried machine;

This arithmetic unit produces the good a plurality of test packets of decision in advance, and is sent to the 3rd network address, and this plurality of packets records the address of this phone port of this second network speech apparatus; And

Receive and analyze the plurality of packets of being passed back by this phone port of this first network speech apparatus that is positioned at this second network address.

11. method for testing speech quality of network speech apparatus as claimed in claim 10, wherein, this network is a LAN.

12. method for testing speech quality of network speech apparatus as claimed in claim 10, wherein, this network is a Wide Area Network.

Images