JP2006222822A - Handover system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To resolve the problem that a temporary interruption occurs at the time of session switching when a speech is continued by switching two or more communication networks in a real-time telephone service. <P>SOLUTION: A handover server which provides a handover function between two terminals is installed, and a structure of relaying a speech session between two terminals by the handover server is introduced. Two sessions of the same speech data are simultaneously established between the handover server and the terminals to continue a speech service without breaks. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to a communication terminal apparatus including receiving means using a plurality of communication paths, and a system including a communication apparatus that provides a telephone handover service to the communication terminal apparatus.

  With the diversification of lifestyles and working styles, there is a growing need to connect to the Internet anytime and anywhere and enjoy communication services using mobile terminals such as mobile phones, PDAs, and notebook computers. Due to recent advances in mobile communication technology, multiple mobile communication networks such as wireless LAN, W-CDMA, cdma2000, and Bluetooth are now available. The mobile terminal can be connected to each communication network by using a communication adapter corresponding to each communication network. Furthermore, by installing multiple types of communication adapters in one mobile terminal, it is possible to select the optimal communication network according to the speed of received data, the communication status of the mobile terminal, etc. It is possible to switch.

  As an Internet communication service using a mobile terminal, IP telephone needs are high, and in recent years it has been used for office extension telephones using wireless LAN. However, at present, an IP telephone is not realized by a public mobile network as a general private telephone. The reason for this is that the public mobile network has a small bandwidth for data communication, and therefore there is a problem that a stable service cannot be provided if the number of terminals accessing one base station increases. In addition, since the public mobile network needs to cover radio waves on a nationwide scale, there is an economic problem such as high communication costs. Therefore, in order to realize the IP telephone service in the mobile body, it is highly useful from the technical and economic viewpoints that the call can be continued by appropriately switching between the public mobile network and the wireless LAN network.

When a mobile terminal is connected to another terminal or server and receives a communication service, when the communication service is dynamically switched and the communication service is continued, the session information of the communication service is transferred to the communication path. Generally, the existing relay device or agent always keeps the communication service in the communication network after switching, and the communication service is continued using this session information.
Here, a session means a communication flow between a transmission device and a reception device that provides a certain communication service, and a set of a series of processes executed in both the transmission device and the reception device to provide the communication service. . Session information is comprehensive information related to a session that provides the service. Examples of session information include the IP address and port number of a server / client terminal, a communication protocol, a session start time, a service elapsed time, a service name, Examples include the application name used.

Usually, the communication service continuation method at the time of switching the communication network is to temporarily suspend the session at the mobile terminal, switch the communication network, and then restart the session. For this reason, in communication services such as video and call voice where the continuity of received data greatly affects the service quality, there is a problem that the user is uncomfortable due to the temporary interruption of the session as described above. Further, there is a problem that it is difficult to continue the same service when characteristics such as the communication speed of the communication network are different before and after switching the communication network.
As a method for solving the difficulty of continuity of service due to the temporary interruption of received data and the difference in communication speed characteristics before and after switching as described above, the session in the communication path before switching and the communication path after switching A method has been proposed in which two sessions are temporarily established simultaneously with a mobile terminal (for example, Patent Document 1).

JP 2004-356922 A

Patent Document 1 provides a method for temporarily establishing two sessions on a communication path before switching and two sessions on a communication path after switching with a mobile terminal. However, Patent Document 1 provides an implementation method that is limited only to a service that distributes stored content, and in a telephone service in which data is transmitted from a client terminal to a server in two sessions, without a temporary interruption. Distribution service cannot be provided.
An object of the present invention is made in view of the above circumstances, and is to provide a method of continuing a telephone service at the time of switching of a communication network in a mobile terminal without being temporarily interrupted.

  In order to achieve the above object, in the telephone handover system of the present invention, a telephone handover terminal (hereinafter referred to as a handover terminal) and a telephone handover server (hereinafter referred to as a handover server) capable of simultaneously establishing a call session with each other in two networks. It is characterized by being configured.

In addition, the handover server in the telephone handover system of the present invention establishes a call session with the call partner of the handover terminal instead of the handover terminal, and transfers call data from the call partner to the handover terminal. Similarly, the handover server transfers call data sent from the handover terminal to the call partner.
And the handover terminal and the handover server in the telephone handover system of the present invention establish two sessions at the same time on the network before and after switching at the time of switching the call path, and send and receive the same call data in both sessions, The session is switched at an appropriate timing based on the time information of the call data received from the two communication paths before and after switching.

As described above, according to the telephone handover system of the present invention, video and sound are temporarily interrupted even in the middle of communication or interactive communication services such as telephone service including video. The user can continue the communication service by switching the communication network.
In the present invention, since the codec and the bit rate can be changed in each session, it is possible to implement a call at a bit rate corresponding to the current throughput in each communication network in a timely manner.
As described above, according to the present invention, even in a plurality of communication networks having different characteristics and a communication network in which the communication speed fluctuates greatly, a bidirectional communication service having a high real-time requirement can be obtained without temporary interruption. Can be provided continuously. Therefore, it is possible to provide a highly convenient service to the user by using a portable terminal that is currently widely used.

  The present application relates to a telephone handover apparatus, a telephone handover terminal apparatus, and a telephone handover system constituted by the telephone handover apparatus for continuing a call without interruption when switching a communication path. Hereinafter, embodiments of the present invention will be described with reference to the drawings.

A telephone handover system according to Embodiment 1 of the present invention will be described with reference to FIGS.
FIG. 1 is a diagram illustrating configurations of a handover server 100 and a handover terminal 105 in the telephone handover system according to the first embodiment. In the figure, a system configuration in the case where the telephone terminal 104 and the handover terminal 105 make a call by relaying the handover server 100 is shown. Here, the telephone terminal 104 is a telephone terminal having a general IP telephone function that does not have the handover function disclosed in the present invention.

  In FIG. 1, the handover server 100 includes a handover processing unit 101, a call processing unit 102, and a communication network interface unit 103. The handover processing unit 101 performs a switching process between a call session via the communication network # 2 (108) and a call session via the communication network # 3 (107), and a call session between the telephone terminal 104 and the handover server 100; The contents of the call in the call session between the handover server 100 and the handover terminal 105 are reflected on each other. The call processing unit 102 controls message transmission / reception and call data transmission / reception in a call session with another terminal. The communication network interface unit 103 is a network interface for accessing the communication network # 1.

  The handover terminal 105 includes a communication network interface unit 109, a call processing unit 110, a handover processing unit 111, and a voice processing unit 112. The communication network interface unit 109 is a network interface for accessing the communication network # 2 (108) and the communication network # 3 (107). The handover terminal can be connected to the communication network # 1 (106) via the communication network # 2 (108) and the communication network # 3 (107). The call processing unit 110 controls transmission / reception of messages and transmission / reception of call data in a call session with another terminal. The handover processing unit 111 performs a switching process between a call session via the communication network # 2 (108) and a call session via the communication network # 3 (107). The voice processing unit 112 encodes a call sound input from an input device such as a microphone and passes it to the handover processing unit 111 (D103a), and also decodes the received call data (D103b) passed from the handover processing unit to generate a speaker or the like. The call sound is output to the output device.

Next, FIG. 2 shows a flow when the handover of the IP telephone service is performed between the handover terminal 105 and the general telephone terminal 104 via the handover server 100. In the figure, an example is shown in which SIP (Session Initiation Protocol: IETF RFC3261) is used as a call control protocol for performing a call for an IP telephone service. However, the handover process of the present invention is performed in accordance with ITU-T H.264. The same applies to other call control protocols such as H.323.
In the figure, the handover terminal 105, the handover server 100, and the telephone terminal 104 directly exchange SIP messages, but the present invention can be similarly applied through a call control server such as a SIP proxy.

  The handover terminal 105 includes a mode flag indicating whether the call processing unit 110 operates with a normal IP telephone function or via a handover server. In the case of the handover server via mode, the call processing unit 110 transmits an INVITE (session establishment request) message (M200) to the telephone terminal 105 to the handover server 100 via the communication network # 2 (108). When the call processing unit 102 in the handover server 100 receives the INVITE message (M200), the call processing unit 102 returns a Ringing (calling) message (M201) to the handover terminal 105 and also includes a destination URI included in the “To” header in the INVITE message. (In the example of the figure, the URI of the telephone terminal 104) is analyzed, and an INVITE message is transmitted to the telephone terminal 104 (M202). When the call processing unit 110 in the handover terminal 105 receives the Ringing message (M201), the call processing unit 110 sounds a ringing tone at the handover terminal 105.

  When the telephone terminal 104 receives the INVITE message (M202), the telephone terminal 104 returns a Ringing message (M203), rings a ringing tone, and waits until the user of the telephone terminal 104 lifts the handset and permits the call. When the call is permitted, an OK message (M204) is transmitted to the handover server 100. When the call processing unit 102 in the handover server 100 receives the OK message (M204), it transmits an OK message (M205) to the handover terminal 105. When the call processing unit 110 in the handover terminal 105 receives the OK message (M205), it returns an ACK (acknowledgement) message (M206) to the handover server 100. Upon receiving the ACK message (M206), the call processing unit 102 in the handover server 100 transmits an ACK message (M207) to the telephone terminal 104. When the telephone terminal 104 receives the ACK message (M207), a call session between the handover terminal 105 and the telephone terminal 104 is established through the handover server 100 via the communication network # 2 (108).

  After this point, the handover terminal 105 transmits and receives call data (D101) to and from the handover server 100. In addition, the handover server 100 transmits and receives call data (D100) with the telephone terminal 104. In the handover processing unit 101 in the handover server 100, the call data (D101b) sent from the handover terminal 105 is stored in the internal buffer area 401, and when the buffer reaches a certain amount, the call data is stored in the oldest order. And the call data (D100a) sent from the telephone terminal 104 is transferred to the handover terminal 105. The handover over terminal 105 stores the received call data (D101a) in the buffer 301, and when the buffer reaches a certain amount, the call data (D101a) is transferred to the voice processing unit 112 in the oldest order (D103). The voice processing unit 112 performs packet analysis of the call data (D103a), voice data separation, and voice decoding, and outputs voice from the output device. Also, voice is input from the input device, voice encoding and packet generation are performed, and the data is sent to the call processing unit 110 as call data (D103b), which is sent to the handover server 100 as call data (D101b). Through the processing as described above, a call between the handover terminal 105 and the telephone terminal 104 is realized. In SIP, the above call data is transmitted and received in a voice packet format according to RTP (Realtime Transfer Protocol).

  Note that FIG. 2 shows an example in which a call is made from the handover terminal 105 to the telephone terminal 104. However, even when a call is made from the telephone terminal 104 to the handover terminal 105, session establishment via the handover server 100 is similarly performed. Done. In this case, the handover terminal 105 needs to perform processing such as registering in advance so as to transfer the incoming call to the handover server 100 (for example, a SIP redirect function). It is necessary for the telephone terminal 104 to be able to know the URI of the handover server 100 as a transfer destination of the handover terminal 105.

Next, in a state where the handover terminal 105 is communicating with the telephone terminal 104 via the handover server 100 as described above via the communication network # 2 (108), the handover terminal 105 somehow communicates with the communication network # 2. 2 (108) When the throughput drop is detected (201), a process of switching the call via the communication network # 3 (107) will be described.
As a method of detecting (201) a decrease in throughput, there is a method of exchanging packet transmission / reception information by RTCP (RTP Control Protocol). In addition, a method of detecting an increase in packet loss from the sequence number of the received RTP packet, a method of detecting a decrease in the remaining buffer capacity, a method of performing dummy communication for throughput measurement, and the like are also applicable.

When the handover terminal 105 detects a decrease in throughput of the communication network # 2 (108) (201), the handover processing unit 110 transmits an INVITE message (M208) to the handover server 100 via the communication network # 3 (107). Upon receiving the INVITE message (M208), the handover processing unit 102 in the handover server 100 receives the incoming caller identifier (using the tag value of the From header, etc.) and the incoming call of the established session via the communication network # 2 (108). It is verified whether the information with the destination address (To header) matches the information in the INVITE message (M208) (202). When it is confirmed by the above collation that the incoming call source and the incoming call destination of communication network # 2 (session established) and communication network # 3 (session establishment requested) match, the call processing unit in handover server 100 102 returns an OK message (M209) to the handover terminal 105. When the call processing unit 110 in the handover terminal 105 receives the OK message (M209), it returns an ACK message (M210) to the handover server 100. At this time, since the session with the handover server 100 via the communication network # 3 (107) is established, the communication network # 2 (108) and the communication network # 3 (107) are simultaneously connected on the two communication networks. Two sessions are established.
Thereafter, the handover server 100 transmits / receives call data to / from the handover terminal 105 (D102). At this time, the handover processing unit 101 in the handover server 100 transfers the same data as the call data already transmitted to the handover terminal 105 to the handover terminal 105 via the communication network # 3 (107) (D102a). The handover terminal 105 continues to store the call data (D102a) in the buffer 302. When the remaining capacity of the buffer 320 is exhausted, the buffer 320 is deleted in the oldest order. The handover processing unit 101 receives call data sent from the handover terminal 105 via the communication network # 3 (107) and stores it in the internal buffer area 402. When the buffer capacity runs out, the old call data is deleted.

  At this time, the call sound (D100) sent from the telephone terminal 104 is sent from the handover server 100 to the handover terminal 105 simultaneously from both the networks # 2 (108) and # 3 (107). . Call data D101 and D102 are sent from the handover terminal 105 to the handover server via the communication network # 2 (108) and the communication network # 3 (107). The call data (D101 of the communication network # 2 (108) is also transmitted. ) Is transferred to the telephone terminal 104 as call data (D100).

  In this state, the handover processing unit 111 of the handover terminal 105 starts executing the handover process 203, and the handover processing unit 101 of the handover server 100 starts executing the handover process 204.

  Here, the handover process 203 performed in the handover terminal 105 will be described. When two call sessions are established as described above, the handover processing unit 111 in the handover terminal 105 receives the call data (D101a) received via the communication network # 2 (108) and the communication network # 3. Call data (D102a) received via (107) is stored in buffers 301 and 302, respectively. In the handover process 203, the time information of the call data stored in each buffer is managed, and the time at which the call data in both buffers is not interrupted in time is determined as the handover time. The subsequent call data (D103a) is switched from the call data (D101a) in the buffer 301 to the call data (D102a) in the buffer 302. However, the difference in time is reduced as much as possible when data that overlaps in time is not stored in both buffers due to a large delay difference between the communication network # 2 and the communication network # 3. Such a time is determined as the handover time, and the call data is switched as described above.

  The handover process 204 performed in the handover server 100 is the same as the handover process 203 (see FIG. 4). That is, at the time when two call sessions are established as described above, the handover processing unit 101 in the handover server 100 receives the call data (D101b) received via the communication network # 2 (108) and the communication network #. Call data (D102b) received via 3 (107) is stored in buffers 401 and 402, respectively. In the handover process 204, the time information of the call data stored in each buffer is managed, and the time at which the call data in both buffers is not interrupted in time is determined as the handover time. The subsequent call data (D100b) is switched from the call data (D101b) in the buffer 401 to the call data (D102b) in the buffer 402. However, the difference in time is reduced as much as possible when data that overlaps in time is not stored in both buffers due to a large delay difference between the communication network # 2 and the communication network # 3. Such a time is determined as the handover time, and the call data is switched as described above.

  When the handover process 204 in the handover server 100 ends, the handover processing unit 101 of the handover server 100 transmits a handover completion message (M211) to the handover terminal 105. The handover terminal 105 knows the end of the handover process 204 of the handover server 100 by receiving the handover completion message (M211). When the handover process 203 and the handover server 204 are completed, the handover terminal 105 transmits a BYE message (M212) to the handover server 100 via the communication network # 2, and the handover terminal 105 transmits to the handover server via the communication network # 2. The transmission of the call data (D101b) to be transmitted is terminated. When the handover server 100 receives the BYE message (M212), the handover server 100 stops transmission of the call data (D101a) to the handover terminal 105 via the communication network # 2, and transmits an OK message (M213) to the handover terminal 100. To do. As described above, by seamlessly switching the communication session from the communication network # 2 to the communication network # 3 by ending the call session of the communication network # 2.

  As described above, in the telephone handover system according to Embodiment 1 of the present invention, a handover server that provides a handover function is installed between two terminals, and a mechanism is introduced in which the handover server relays a call session between the two terminals. The same call data session can be simultaneously established between the handover server and the handover terminal. Each of the handover terminal and the handover server acquires the reproduction time of the two call data received in these sessions, determines the timing so as not to cause interruption as much as possible, and switches both data. Therefore, the seamless handover function that can be realized only by the conventional one-way distribution service can be realized also by a bidirectional communication service such as an IP telephone.

  In a second embodiment of the present invention, as a modification of the first embodiment, when the handover processing unit 111 of the handover terminal 105 sends the call data D103b passed from the voice processing unit 112 to the handover server, A configuration is adopted in which the call data D103b is sent to the handover server 100 after reaching a certain buffer amount. Similarly, when the handover processing unit 101 of the handover server 100 sends the call data D100a sent from the telephone terminal 104 to the handover terminal 105, it accumulates in the buffer and then sends it to the handover terminal 105 after reaching a certain buffer amount. Take the configuration. According to the second embodiment of the present invention, the handover terminal 105 and the handover server 100 both reduce the influence of rate fluctuation and jitter at the time of packet transmission to the communication network by providing a certain delay in the buffer at the time of transmission. Can do.

  The third embodiment of the present invention has a configuration in which the codec and the content bit rate are changed according to the monitored throughput as a modification of the second embodiment. According to Embodiment 3 of the present invention, packet loss can be reduced by adapting the codec and bit rate at the time of transmission to the throughput.

  The fourth embodiment of the present invention has a configuration including not only audio but also a video processing function as a modification of the first embodiment. A seamless handover function can also be realized for the IP-TV telephone service by adding the same configuration as the call data of the first embodiment to the video data.

1 is a block diagram showing a configuration example of a telephone handover system according to Embodiment 1 of the present invention. The figure which shows the example of the processing flow of the telephone handover system by Embodiment 1 of this invention. The figure which shows the structural example of the hand-over process part in the hand-over terminal of the telephone hand-over system by Embodiment 1 of this invention. The figure which shows the structural example of the handover process part in the handover server of the telephone handover system by Embodiment 1 of this invention.

Explanation of symbols

100: Telephone handover server 101: Handover processing unit (server)
102 ... Call processing unit (server)
103 ... Communication network interface unit (server)
104 ... Telephone terminal 105 ... Handover terminal 106, 107, 108 ... Communication network 109 ... Communication network interface unit (terminal)
110: Communication processing unit (terminal)
111 ... Handover processing unit (terminal)
112: A voice processing unit (terminal).

Claims (2)

It is a handover server that can talk to one terminal simultaneously on two or more communication paths, and can talk to two terminals on one or more communication paths,
A call processing unit for establishing a call with the one terminal on the two or more communication paths;
Call data sent from the two terminals is simultaneously transferred to two or more communication paths of the one terminal, and call data received from any of the two or more communication paths of the one terminal is stored in a buffer. A handover server having a handover processing unit for transferring to the above two terminals,
When switching the communication path of call data to be transferred to the second terminal, the handover processing unit seamlessly sets the call data communication path based on time information of call data simultaneously received from the two or more communication paths. A handover server, wherein the handover server is switched and forwarded to the above-mentioned 2 terminal. A communication network interface unit capable of simultaneously establishing two or more communication networks with the handover server according to claim 1;
A call processing unit for simultaneously establishing two or more calls with the handover server via two or more communication paths via the two or more communication networks;
A handover processing unit that simultaneously transmits transmission voice data to the two or more communication paths and stores call data received from the two or more communication paths in a buffer;
A telephone terminal having a voice processing unit for decoding and reproducing call data received from the handover processing unit,
When the handover processing unit switches the communication path of the call data received from the handover server, the communication path of the call data received without interruption based on the time information of the call data received simultaneously from the two or more communication paths. A telephone terminal characterized by switching.

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