KR100395680B1 - Method for controlling auto connection conversion - Google Patents

Abstract

A method for automatically switching to a call between a regular phone and an internet phone is disclosed. The method for automatically switching to a call between a general telephone and an internet phone of the present invention includes: a first step of counting the time and number of data packets being input for a predetermined time by driving a timer during a call through the internet phone; A second step of calculating an error rate of packet loss generated when transmitting a packet for real-time communication on the Internet from the first step; A third step of searching for a general telephone number associated with an IP when the error rate is equal to or greater than a set value; And a fourth step of making a call connection to the general telephone number. According to the present invention, in case of a call failure according to the network state, the call is automatically attempted from the Internet phone to the normal phone, so that the user can not only provide continuous calls but also terminate the Internet phone and reconnect the call to the normal phone. Eliminate the inconvenience of trying. As a result, when using the Internet phone, there is an effect of smoothing the user's telephone call by minimizing the discomfort caused by the failure of the network connection.

Description

How to automatically switch to landline phone and internet phone call {Method for controlling auto connection conversion}

The present invention relates to a method for automatically switching to a call between a general telephone and an internet phone, and in particular, detects a poor call quality according to a network condition when talking on an internet phone, and when the call quality is poor, matches the IP with a general phone number. The present invention relates to a method of automatically switching to a call of an ordinary phone and an internet phone to automatically switch a call path.

Recently, Internet phones have presented a new paradigm of the communication revolution. There are many ways of Internet phone, from the most basic PC to PC method to PC to Phone method and Phone to Phone method. Internet phones are almost the same in principle as regular phones. In the landline, when the receiver picks up the handset and presses the telephone button, the signal goes through the telephone company's relay to find the number of the registered party. When the voice data is converted into an electrical signal and transmitted through a copper wire, the voice data is converted into voice data through a nearby telephone station or a base station and transmitted to the sender.

Internet phones have the same principle. In the PC-to-PC method, which is the most basic form, when the user executes the Internet phone-related software and presses the Internet phone address of the other party, the signal acts as a telephone company, and the Internet phone server acts as a relay between the receiver and the sender. Find the other party through. An Internet phone address is similar to a phone number and is registered on the Internet phone server. Speaking of the voice data, the voice data is compressed and converted into an Internet standard signal, which is transmitted through an Internet network through an Internet phone server, and then converted into voice again by the other party's PC. In this case, when the voice signal is transmitted and received using an Internet phone through a network, the voice is binary-coded using the 'Voice Codec' technology, which compresses and restores the voice to be as small as possible. When compressed and transmitted, the other party recovers the voice data and then decodes the encoded voice. Currently, G.723 is used at 5.3Kbps and 6.3Kbps, and G.729 is used at 8Kbps.

In addition, a protocol that can effectively transmit voice data over the Internet plays an important role, and H.323, a standard protocol for audio and video communication over a LAN that does not guarantee service quality, and to improve the sound quality of restored voice data RTP / RTCP protocol provided in the packet, and RSVP protocol is used to reserve resources such as bandwidth along the path to satisfy the quality of service (QoS) to provide.

On the other hand, PC to Phone, which is a step forward from PC to PC, is designed to communicate with each other by connecting the internet network and the existing telephone network. It is the Gateway that was developed to make this possible. Here, the gateway is a kind of server program, and serves as a bridge connecting the existing telephone network, that is, public switched telephone network (PSTN) and the Internet. The gateway functions used in PC to Phone can be classified into four categories. Specifically, ① the function of making a call to the other party by generating DTMF Tone, ② the function of sending a voice to the telephone, ③ the function of checking the phone state to detect the busy signal and confirm that the other party hangs up, and ④ the telephone There is a function to break.

Now, a call process using a conventional internet phone will be described.

1 is a flowchart schematically illustrating a process of performing a call using a conventional Internet phone. As shown in FIG. 1, after initializing a system by applying power to a user's PC in order to access the Internet (S10), an attempt is made to access a terminal server of an ISP. When a connection is made to a terminal server of an ISP, data can be exchanged. PPP is operated to obtain an IP address from the server. As soon as the IP address is assigned, TCP / IP is ready to run so that various Internet programs can be run to connect with hosts (computers connected to the Internet) on the Internet.

Then, the user executes the Internet phone-related program (S12), and attempts to connect by inputting a phone number or IP to the executed Internet phone-related program (S14). When the counterpart answers (S16), the call is made (S18). At this time, in general, the Internet phone is used by using a protocol (H323, SIP, MGCP, etc.) that can be used in the network. Here, the h323 is the latest of the H.32X series of recommendations that define standards for video conferencing on various types of networks, and is derived from H.320, a video conferencing standard on ISDN and circuit-switched networks. H.323 was developed to extend the multimedia service target network to the general packet switched network including the corporate intranet. The main purpose of H.323 is to maintain interoperability through interworking with other multimedia service networks. In the field of IP telephony, such as Internet phones and Voice over IP (VoIP), H.323 is used because of the interoperability on the different networks that H.323 supports. This H.323 has network independence, bandwidth management, conferencing between networks, platform and application independence, multipoint support, multicast support, interoperability, codec standards, and flexibility.

On the other hand, the error is checked during the call (S20) to determine whether an error has occurred (S22), while the network problem occurs during the Internet phone call, the adaptive differential pulse code modulation (Adaptive Differential Pulse Code Modulation) ADPCM) If the data packet cannot be received within the given time, the call quality or delay of the Internet phone is displayed. As described above, when the telephone call is connected using the network, when using the G723.1 protocol, the transmission speed in the network has a transmission speed of 5.3 Kbps or 6.3 Kbps. In the case of 5.3 Kbps, 20 bytes of ADPCM data is 30ms. In case of 6.3Kbps, 24byte ADPCM Data should be received in 30ms. That is, during the Internet phone call, if the ADPCM sent from the network is not received every 30ms, the call quality is abnormal.

However, even if the call quality is poor, the error is corrected (S24) and the error is recovered (S26). In other words, the channel continues to be maintained until the user disconnects the channel.

On the other hand, if error recovery is not made, the connection is disconnected (S28).

Therefore, in the prior art, even if the call quality is poor, the telephone call is continued by maintaining the call path in a state where the call quality is poor, so that the user cannot actively cope with communication or communication with the caller not being made correctly or disconnected. There was discomfort.

In addition, if the continuous bad phone call continues, there was an uncomfortable problem of having to terminate the Internet phone, try to call the other party using a separate phone.

Accordingly, an object of the present invention is to detect a poor call quality according to the network condition when the Internet phone call, and when the call quality is poor, the normal phone and the Internet phone to automatically switch the call path by the matching of the IP and the general phone number It is to provide a method for automatic conversion to a call.

More specifically, when there is a sudden increase in traffic on the network and various other causes, the ADPCM Data Packet cannot be continuously received within a given time, or when the call quality is poor or the network is disconnected. Use the IP information used to connect to find the phone number in the built-in address book, or find the phone number registered in the Gate Keeper, and then provide the user with information about the quality of the Internet phone and the phone number. In addition, when the user wants to connect to a regular phone, it provides a method for automatically switching to a call between an ordinary phone and an internet phone that terminates the Internet phone and connects to the general phone number.

1 is a flowchart schematically illustrating a process of performing a call using a conventional Internet phone;

2 is a control block diagram schematically showing the configuration of an Internet phone according to an embodiment of the present invention;

3A to 3C are flowcharts illustrating a process of automatically converting an Internet phone call into a call to a normal phone according to an embodiment of the present invention.

Explanation of symbols on the main parts of the drawings

100: sender system 200: receiver system

201: Initial value setting part 202: Auto call forwarding setting / release part

203: Dial-up Software 204: Operating System

205: switch control unit 206: network connection unit

207: general telephone connection 208: handset

209: Speaker and Microphone 210: Regular Phone

The present invention for achieving the above object of the present invention relates to a method for automatically switching to a telephone call of an Internet phone and a phone of the Internet phone, which stores its own phone number and is connected to a common gatekeeper, and the general of the present invention. A method for automatically switching to a call between an internet phone and a phone includes: a first step of counting the time and number of data packets being input for a predetermined time by driving a timer during a call through the internet phone; A second step of calculating an error rate of packet loss generated when transmitting a packet for real-time communication on the Internet from the first step; A third step of searching for a general telephone number associated with an IP when the error rate is equal to or greater than a set value; And a fourth step of making a call connection to the general telephone number.

At this time, it is preferable to further perform the step of displaying and alerting the user of an error occurrence between the third and fourth steps.

In addition, the packet loss is any one selected from delay, jitter, loss amount, and loss pattern, and the packet is data that performs adaptive differential pulse code modulation (ADPCM) transmitted from the other party due to a problem on the network during an internet phone call. Packet.

In the second step, an error rate is calculated through delay and loss using a timestamp and a sequence number defined in a real-time transmission protocol standard (RTP).

The third step may include: searching for a general telephone number associated with an IP stored in the Internet phone; Searching for a gatekeeper next if there is no general telephone number associated with the IP; And stops error checking when the general phone number associated with the IP is not found and displays the general phone number on the Internet phone when the general phone number associated with the IP is found from the Internet phone or the gatekeeper. Consists of steps.

In the present invention, it is based on RTP for measuring the packet loss rate and loss pattern and the temporal characteristics (or delay characteristics) such as end-to-end delay, jitter, and out-of-order packets.

In real-time data transmission over the Internet, temporal (or delayed) characteristics such as end-to-end delays, jitter, out-of-order packets, and loss characteristics such as packet loss rates and loss patterns It is a very important factor. As a result, delay and loss are achieved by a timestamp and a sequence number defined in the Real-time Transport Protocol (RTP), a network simulator that can measure real-time data transmission characteristics on the Internet. Can be measured. This method can be very useful in finding a suitable trade-off value of recovery rate that varies with delay and transmission rate, and can also be used as a simulator for packet recovery algorithm development.

RTP is extended in relation to client and server, and it allocates a random time (for example, 1 second) for each packet size, adjusts the number of packets, and transmits packets at the same transmission rate for each packet size. By counting the packets received in the packet it is possible to know the packet reception rate for each packet size.

On the client side, the packet recovery method, audio codec, redundant audio codec, packet size, frames per packet, interleaving method, and loss are inputted. From this information it is possible to determine the packet size and transmission interval, and ask the server for this value. The server transmits a packet to the client at a predetermined packet size and transmission interval from the received information. At this time, the packet to be transmitted includes a timestamp and sequence number information. The time stamp defined by RTP represents the sampling time at which real-time data is generated and has a size of 32 bits. The upper 16 bits can measure 18 hours in seconds, and the lower 16 bits have a resolution of 15 ms. A sequence number having a size of 16 bits is a serial number assigned to each transport packet and used to measure a packet loss amount and a loss pattern.

Hereinafter, with reference to the accompanying drawings will be described a preferred embodiment of the present invention.

In the present invention, various various embodiments may be presented, but a description will be given of a process of switching from an internet phone (sender) to a general phone (receiver) when a call connection is made to an internet phone (receiver) through an internet phone (sender). Shall be. Of course, it is well known that the sender can also switch to a regular telephone.

2 is a control block diagram schematically showing the configuration of an Internet phone according to an embodiment of the present invention. In order to implement a method for automatically switching to a call between a general telephone and an internet phone of the present invention, as shown in FIG. 2, a sender system (1) having a system configured to use an internet phone and a general telephone and an internet phone are selectively provided. It should be largely composed of a receiver system 200 configured to use the system.

The sender system 100 is a system that can be connected to the normal Internet and includes a program related to the Internet phone, and thus a detailed description thereof will be omitted.

On the other hand, the receiver system 200 in detail the initial value setting unit 201, automatic call forwarding setting / canceling unit 202, dial-up software 203, operating system 204, switch control unit 205, network The connection part 206, the general telephone connection part 207, the handset 208, the speaker and the microphone 209, and the general telephone 210 are comprised.

The initial value setting unit 201 sets the phone number of the telephone exchange that performs call forwarding, and the phone number of the ISP to which call forwarding is performed, and whether the call forwarding is set automatically or manually. Also set. The automatic call forwarding setting / release unit 202 sets the operating system 204 when the user connects to the Internet service provider (ISP) using the PPP of the dial-up software 203 while the initial value for the forwarding call is set. Detects this and executes the automatic call forwarding setting by the automatic call forwarding setting / release unit 202.

The dialing software 203 is a program for connecting to the Internet and making a call. The operating system 204 has a program for driving the system as a whole according to the switching mode state of the Internet phone or a regular phone. The switch control unit 205 refers to a general electrical switch in which switch control is selectively performed according to a switching mode state of an Internet phone or a general phone. The network connection unit 206 is composed of a modem or a network adapter configured to connect to the Internet or a network. In particular, the network adapter preferably uses 10 Mbps Ethernet for high speed communication. The network may be more preferably a high speed network such as a fiber base 100 Mbps or ISDN line. The general telephone connection portion 207 means a normal communication line connection portion where a PSTN connection is made. The handset 208 is configured to be connected to the switching control unit 205 so as to selectively communicate with an Internet phone or a regular phone. The speaker and the microphone 209 form a normal headset. The general telephone 210 means a conventional telephone which is well known.

Although not shown in the drawings of the present invention, an exchange server (not shown) is placed between the general telephone connection unit 207 and the general telephone 210 so that an internet server (not shown) and a gateway (not shown) are connected. These Internet servers (not shown) and gateways (not shown) are connected to the Internet network (not shown). The network connection unit 206 and the sender system 100 are also connected to an internet network (not shown).

It describes the operation process of the configuration necessary to implement the automatic switching method of the call of the general telephone and the Internet phone of the present invention configured as described above.

3A to 3C are flowcharts illustrating a process of automatically converting an Internet phone call into a call to a normal phone according to an embodiment of the present invention. As shown in Figure 3a, after the power to the user's PC to initialize the system to access the Internet (S100), attempts to access the terminal server of the ISP. When a connection is made to a terminal server of an ISP, data can be exchanged. PPP is operated to obtain an IP address from the server. As soon as the IP address is assigned, TCP / IP is ready to run so that various Internet programs can be run to connect with hosts (computers connected to the Internet) on the Internet.

Then, the user executes the Internet phone-related program (S110), and attempts to connect by inputting a phone number or IP to the executed Internet phone-related program (S120). When the counterpart responds (S130), the call is made (S140). At this time, the present invention also uses the Internet phone using a protocol (H323, SIP, MGCP, etc.) that can use the telephone in the network.

Then, the call sensitivity is checked (S150), the specific process of which is as follows. As shown in Figure 3b, by driving a 1000ms timer (S151) counting the time the ADPCM Data Packets incoming and the number of ADPCM Data Packets for a predetermined time (20byte * (1000ms / 30ms) in the case of 5.3Kbps); 6.3 In the case of Kbps, 24 bytes * (1000 ms / 30 ms) (S153), and the error rate is calculated based on this counting value (S155) to determine whether the call quality.

On the other hand, it is determined whether the call sensitivity error rate is greater than the set value according to the calculation of the error rate (S160). As a result of the error rate determination, if it is determined that the call sensitivity error rate is larger than the set value, the general telephone number associated with the IP is searched (S170). At this time, in searching for a general phone number corresponding to the IP address of the other party on the call, find a file in the address book DB, which is an application that the Internet phone itself has, or from a gatekeeper when interlocked with a gatekeeper. Can be.

As shown in FIG. 3C, the specific process first searches for a general telephone number associated with an IP stored in the Internet phone (S171). On the other hand, it is determined whether there is a general telephone number associated with the IP in the Internet phone (S172). If the general telephone number is not searched, the gatekeeper is searched next (S173). On the other hand, it is determined whether there is a general telephone number associated with the IP (S174). If the general telephone number is not found, the error checking is stopped (S175) and the process proceeds to step S140.

On the other hand, when a general telephone number is found in each of the steps S172 and S174, the retrieved general telephone number is displayed on the Internet phone (S176).

Here, the error checking operation is stopped even when the user refuses to connect to the general telephone. The gatekeeper is a program that manages, controls, and finally approves a system. In addition, the address book has at least 200 lists, and can store four telephone numbers (general telephone number or IP) per list, and are stored in the form of a file. The method of searching for a general phone number in the address book using the IP informs the user of the first general phone stored in the remaining three phone numbers when an IP matching the IP information in the address book file can be found.

On the other hand, after retrieving the general telephone number through the above process, it is determined whether the error occurs and the PSTN access to the user (S180). If the user does not request the PSTN connection, the process proceeds to the step S140 and continues the Internet phone call. If the user requests the PSTN connection (S190), the user disconnects the Internet phone and requests the call to the general phone. It is made (S200). After the call connection is made (S210) and the call is made through the PSTN (S220), when the user ends the call, the PSTN connection is disconnected to terminate the call (S230).

On the other hand, if it is determined in step S160 that the call sensitivity error rate is not greater than the set value, the error is recovered (S240), and the error recovery is completed (S250). Continues, and if error recovery is not completed, the flow proceeds to step S170.

As described above, the method for automatically switching to a call between the normal phone and the Internet phone according to the present invention, when a call failure according to the network state automatically attempts to connect the call from the Internet phone to the normal phone to provide a continuous call to the user In addition, it can eliminate the inconvenience of shutting down the Internet phone and retrying the call. As a result, when using the Internet phone, there is an effect of smoothing the user's telephone call by minimizing the discomfort caused by the failure of the network connection.

The present invention is not limited to the above-described embodiment, and it will be apparent that many modifications are possible by those skilled in the art within the technical spirit of the present invention.

Claims (6)

In the method of automatically storing the telephone number, and automatically switching to the telephone call of the Internet phone and the Internet phone connected to the normal gatekeeper, A first step of counting the time and number of data packets being input for a predetermined time by driving a timer during a call through an Internet phone; A second step of calculating an error rate of packet loss generated when transmitting a packet for real-time communication on the Internet from the first step; A third step of searching for a general telephone number associated with an IP when the error rate is equal to or greater than a set value; And A fourth step of making a call to the general telephone number Method of automatic switching to the call of the general phone and Internet phone, characterized in that made. The method of claim 1, further comprising displaying and alarming an error occurrence to the user between the third and fourth steps. The method of claim 1, wherein the packet loss is any one selected from delay, jitter, loss amount, and loss pattern. [4] The general telephone and the internet telephone according to claim 1 or 3, wherein the packet is a data packet which performs an adaptive differential pulse code modulation (ADPCM) transmitted from the other party due to a network problem during an internet phone call. How to automatically switch to calls. The method as claimed in claim 1, wherein the second step calculates an error rate through delay and loss using a timestamp and a sequence number defined in a real-time transmission protocol standard (RTP). How to automatically switch to the call of the Internet phone. The method of claim 1, wherein the third step, Searching for a general telephone number associated with an IP stored in the internet phone; Searching for a gatekeeper next if there is no general telephone number associated with the IP; And Stopping the error check when the general phone number associated with the IP is not found and displaying the general phone number on the Internet phone when the general phone number associated with the IP is found from the Internet phone or the gatekeeper; Method of automatic conversion to the call of the general phone and Internet phone, characterized in that consisting of.