JP4684240B2 - Multi-person call terminal, multi-person call system, multi-person call method, multi-person call program and recording medium storing program - Google Patents

Description

  The present invention relates to a multi-party call technology that uses a IP communication based on a multicast communication method and does not use a call control server.

  As a new usage of mobile phones, a technology called Push-to-Talk has been developed. Push-to-talk is a technology for realizing a conversation style like a transceiver that “presses” a button and “speaks”. In general, a telephone can only make a one-to-one call, whereas a push-to-talk call can make a one-to-N (many) call, and can simultaneously transmit voice to a plurality of parties. Push-to-talk is formally called Push-to-Talk over Cellular, called “PoC” for short, and is being standardized by the industry organization OMA (Open Mobile Alliance).

  Push-to-talk is one of the so-called “VoIP” s that use the Internet protocol to make a call over a data communication network, and has been developed for mobile terminals such as mobile phones and PDAs. It has already spread mainly in the United States, and in Japan, several mobile phone manufacturers are developing products with the same functions from mobile operators.

  In order to make a call using the Internet, a Session Initiation Protocol (SIP) is used. SIP is a protocol for establishing a real-time association called “session” in order for an IP telephone connected to an IP network to make a call. Standardized by RFC3261 and other regulations. An example of a conventional SIP is shown in FIG. As shown in FIG. 5, when establishing a SIP session, it is common to specify the address of the terminal on the called side from the terminal on the calling side.

FIG. 6 is a diagram illustrating a conventional multi-party call system. As shown in FIG. 6, the conventional multi-person (group) call system requires a device for managing calls called a server. In general, a server is an expensive device and a device that requires skill in setting and maintenance, and there has been a demand for a system that can realize a multi-party call easily at a lower cost. Further, the conventional multi-party call system has a problem that the network load fluctuates due to increase / decrease in call participants. For this reason, a stable call environment has been realized by, for example, securing a call band in advance, but this is not always desirable from the viewpoint of network operation efficiency (Non-Patent Documents 1 to 3).
http://pr.fujitsu.com/jp/news/2004/06/29-2.html http://itpro.nikkeibp.co.jp/article/KEITAI/20051026/223557/ http://www.tdk.co.jp/techmag/knowledge/200601u_a/index.htm

  Accordingly, an object of the present invention is to provide a communication terminal, a multi-person call system, and a multi-person call method that can make a multi-person call without going through a server.

The present invention for solving the above problems relates to the following items.
[1] A call terminal that performs a call over an IP network based on a multicast communication method, and requests a call to the self terminal based on an IP network connection unit, a call unit, a display operation unit, and a call protocol A call terminal for multi-party calls, comprising: a call session control unit that establishes a call session by performing a call.

[2] The call protocol uses a session start protocol (SIP), the destination address is the same as the source address, and includes an item for specifying a multicast address in the connection information of the content part The telephone terminal according to [1].

[3] A program for operating a session start protocol (SIP) for operating the call terminal according to [1], wherein a transmission destination address and a transmission source address are the same, and multicast is added to connection information of a content part. A program including an item for designating an address.

[4] A computer-readable recording medium storing the program according to [3].

[5] An IP network and two or more call terminals according to claim 1 connected to the IP network, and each call terminal makes a call request to the self terminal based on a call protocol. A multi-party call system characterized in that a call session is established and a call is started from any one of the call terminals that has established the call session to another call terminal.

[6] The call terminal according to [5], wherein the call terminal starts a call from another call terminal without performing transmission right control after the call from any one of the call terminals is ended. Multi-party call system.

[7] The multi-party call system according to [5] or [6], wherein the system does not include a server that controls a call between the call terminals.

[8] The multi-party call system according to any one of [5] to [7], wherein the system transmits multimedia information other than sound such as images simultaneously with a call.
[9] The multi-party calling method according to any one of [5] to [6],
(S1) Each of the call terminals establishes a call session by making a call request to the self terminal based on a call protocol,
(S2) A call signal is transmitted from one of the call terminals that has established a call session to the other call terminal,
(S3) A multi-party call method characterized in that another call terminal receives the call signal and thus makes a call between the call terminals.

[10] Furthermore,
(S4) The multi-party call method according to [9], wherein after the call is finished, a call is made from another call terminal without controlling the transmission right.

According to the present invention, an inexpensive multi-party call can be realized by eliminating the need for a server.
In addition, by adopting Multicast as the communication method, a multi-party call method in which the network load does not depend on the number of call participants can be realized.

(Overall explanation)
The present invention adopts a multicast method as a communication method, distributes one voice data to all call participants, connects a SIP Session to itself, and establishes a session by itself. Based on the technical idea that it is possible to eliminate the need for a server for distributing the mobile phone to all terminals and, at the same time, to realize a call environment with a constant load regardless of the number of call participants. It is. Hereinafter, embodiments of the present invention will be described with reference to the accompanying drawings.

(Call terminal)
First, a call terminal for multi-person calls according to the present invention will be described with reference to FIGS. FIG. 1 is a diagram showing a multi-party call using the multi-party call terminal of the present invention, FIG. 2 is a diagram showing a configuration of a call terminal for multi-party call of the present invention, and FIG. It is drawing which shows the telephone call system using the telephone terminal for multi-person calls of invention.

  The call terminal for multi-party calls according to the present invention is configured from a terminal capable of making a call over an IP network based on a multicast communication method, and is not particularly limited as long as it is such a terminal. It can be a PDA, a notebook personal computer, a mobile phone or the like that is possible and has voice input / output means. Furthermore, for example, it may be configured from a terminal connected to a personal computer system via an interface such as a USB.

  Specifically, the communication terminal 10 of the present invention includes an IP network connection unit 11 for performing processing for connecting to an IP network, and a call session control unit for performing connection control for performing a call using the Internet. 12, a display operation unit 13 for performing a call operation and displaying a call state and the like, and a call unit 14 for inputting a voice from the user and outputting a voice to the user. .

The IP network connection unit 11 includes an IP packet unit 15 that converts a digital signal into an IP packet for transmission to the IP network, and conversely restores an IP packetized signal received from the IP network into an original digital signal. Have.
The display operation unit 13 has various operation buttons for controlling connection / disconnection to / from the IP network and transmission / reception, and transmits the user's button operation information to the call session control unit 12. It has a display such as an LED for displaying a call state or the like based on information from the control unit.
The call session control unit 12 performs connection control processing to make a call using the Internet based on the button operation information from the display operation unit 13. Then, the state is transmitted to the display operation unit 13.
The call unit 14 includes a voice input unit 16 that inputs a user's voice as an analog signal through a microphone, a coder unit 17 that compresses the input analog signal into a digital signal, and another call terminal. Received from the IP network connection unit 11 and decoded by the decoder unit 18 for decoding the digital signal into an analog signal, and the audio output unit 19 for outputting the decoded analog signal to the user from a speaker or the like, Is mainly composed of
There are several types of coder means 17 and decoder means 18 of the call unit 14, and it may be necessary to match the call terminal of the other party. In this case, information received from other call terminals is used as a call. The session control unit 12 extracts the information and notifies the communication unit 14 of the extracted information. The communication unit 14 can select and switch between the appropriate coder unit 17 and the decoder unit 18.

SIP
Establishing a session requires a connection requesting party and one that receives the connection request, but when establishing a SIP session for a multicast address, there is no one that receives a connection request and the establishment of a SIP session. However, in the present invention, the IP network connection means 11 connects the SIP Session to itself and establishes the Session alone by establishing the Session. It is characterized by establishing a session.

  More specifically, the SIP session is established by a program as shown in Table 1 below.

  As shown in Table 1, in the present invention, a call session (SIP session) is established by designating the address of its own terminal. Table 1 is a SIP format that describes the contents of a multimedia session. The content part is described in SDP (Session Description Protocol), described in the form of “type = value”, and SDP is defined in RFC4566.

  The present invention is characterized in that a multicast address is specified in the format of “C = IN IP4 224.0.0.10” in the SDP with 1. invite signal and 4. 200 OK signal. The multicast address numbering system is defined in RFC1060. In Table 1, it is also a feature of the present invention that to and from, that is, the destination address and the source address match in all signals.

  By doing so, it is possible to talk in the same way as a transceiver without violating the conventional SIP message, and multiple multicast addresses can be selected, and arbitrary addresses are used in groups. By doing so, the function corresponding to the channel of the analog transceiver can be realized.

  Such a program is new. Therefore, the present invention is characterized in that the session start protocol (SIP) is used, the destination address and the source address are the same, and the item of specifying the multicast address is included in the connection information of the SIP content part. A program for making a multi-party call over an IP network based on a multicast communication system (for providing a call terminal with a transceiver function through an IP network) is also within the scope of the present invention.

  In the present invention, the session start program is not limited to SIP, but can be similarly applied to IP telephone systems other than SIP, such as Arrowfone of Japan Systemware, Skype of Skype, and a call protocol.

(Multi-party call system)
Next, a communication method using the call terminal of the present invention will be described.
The call system of the present invention is a system for performing multi-party calls with at least two call terminals as shown in FIG. 2 in an IP network system as shown in FIG.

The IP network system can be a conventionally known network system. However, in the present invention, a closed IP network system in a company, a factory, a facility, or the like is preferable because a caller can be specified.
However, the present invention is not limited to these closed systems, and can be widely applied to, for example, multi-party calls between groups based on predetermined user authentication.
Authentication by such an IP network system is realized by a method well known in the art.

(Operation of multi-party call system)
Next, the operation of the multi-party call system of the present invention will be described with reference to the drawings.
FIG. 4 is a sequence diagram showing a call flow of the multi-party call system according to the present invention.

  As shown in FIG. 4, in the multi-party call system of the present invention, as described above, each call terminal establishes a call session by making a call request to its own terminal based on the call protocol (initialization / standby). ).

Next, the calling-side call terminal transmits (sends) a call signal to the called-side terminal. Other call terminals receive (receive) the call signal.
In such a state, a call is performed between the terminal on the transmission side and the terminal on the reception side. Disconnect after the call ends.

  In the establishment of the conventional SIP session shown in FIG. 5, the address of the receiving terminal is specified from the calling terminal, but in the present invention, the SIP session is established by specifying the address of its own terminal.

  Further, in the multi-person call system of the present invention shown in FIG. 4, the transceiver function in the call terminal has a feature that the user can activate at an arbitrary timing. That is, in the conventional multi-person call system shown in FIG. 5, synchronization of SIP signals with other terminals is required, whereas in the call system of the present invention, synchronization of SIP signals with such other terminals is not possible. The point that becomes unnecessary is different.

  Therefore, it is possible to embody a multi-party call without complicated operation for such synchronization, that is, without performing “transmission right control”. In addition, since it can be activated at any timing of the user, it is possible to freely make a multi-party call between users.

(Preferred operation example)
In the multi-person call system according to the present invention, it is preferable that the exchange of the transmission right is performed by the user of the call terminal.
In other words, in the case of the conventional push-to-talk implementation, after establishing the SIP Session, “until you decide who has the microphone, who wants to hold the microphone, and who gives the microphone to” Need to control "speaking right". Such control of “transmission right” is realized by processing between the server and the client (the mobile terminal of the call participant). Such a “sending right” control process becomes very complicated, and the traffic for the process may increase greatly.

In contrast, the preferred embodiment of the present invention is a multi-party call system that does not use a server, and does not perform a process between a conventional server and a client.
As a result of intensive studies by the present inventors, when the multi-party call system of the present invention is applied and the transmission of the “send right” is left to the call participant's judgment, that is, “the conventional analog transceiver When it was executed in the same manner as the “operating method”, it was found that it was possible to make a multi-party call at a sufficiently practical level.

In other words, the multi-party call system of the present invention is extended to a system that does not have a server that controls the “transmission right”. Similarly, the multi-person call method of the present invention is also extended to a multi-person call method for making a call between a large number of people without controlling the “transmission right”.
However, the present invention is not limited to these.

The call unit shown in FIG. 4 is also performed by IP packet communication, and a multicast address is designated as a transmission destination address, so that a plurality of call terminals designated by one transmission can receive a call. This also reduces communication traffic on the IP network.
In the present embodiment, a voice call has been described as an example, but it goes without saying that transmission of various multimedia information such as image information can be handled.
Therefore, the multi-person call system (call method) of the present invention is extended to a multi-media system that exchanges multimedia such as voice and images among a large number of people (between a plurality of terminals).

It is drawing which shows the multi-party call using the multi-party call terminal of this invention. It is drawing which shows the structure of the call terminal for multi-person calls of this invention. It is drawing which shows the telephone call system using the telephone terminal for multi-person calls of this invention. It is a sequence diagram which shows the multi-person call system call flow of this invention. It is a sequence which shows an example of the conventional SIP. 1 is a diagram illustrating a conventional multi-party call system.

Explanation of symbols

10, 10 ', 10 "terminal 11 IP network connection means 12 communication session control unit 13 voice input means 14 voice output means 15 display means

Claims (10)

A call terminal that performs a call over an IP network based on a multicast communication method,
An IP network connection unit, a call unit, a display operation unit, a call session control unit that establishes a call session by making a call request to the self terminal based on a call protocol;
A call terminal for multi-party calls, characterized by comprising:   The call protocol is a session start protocol (SIP), wherein a destination address and a source address are the same, and includes an item for specifying a multicast address in connection information of a content part of SIP. Item 4. The call terminal according to Item 1.   A program for operating a session start protocol (SIP) for operating a telephone terminal according to claim 1, wherein a transmission destination address and a transmission source address are the same, and a multicast address is included in connection information of a content part of SIP The program characterized by including the item which specifies.   A computer-readable recording medium in which the program according to claim 3 is stored. An IP network and two or more call terminals according to claim 1 connected to the IP network,
Each of the call terminals establishes a call session by making a call request to the self terminal based on a call protocol,
A multi-party call system, characterized in that a call is started from any one call terminal that has established a call session to another call terminal.   The multi-party call according to claim 5, wherein the call terminal starts a call from another call terminal without performing transmission right control after the call from the one call terminal is ended. system.   The multi-party call system according to claim 5 or 6, wherein the system does not include a server for controlling a call between the call terminals.   The multi-party call system according to any one of claims 5 to 7, wherein the system transmits multimedia information other than voice simultaneously with a call. The multi-party calling method according to any one of claims 5 to 6,
(S1) Each of the call terminals establishes a call session by making a call request to the self terminal based on a call protocol,
(S2) A call signal is transmitted from one of the call terminals that has established a call session to the other call terminal,
(S3) A multi-party call method characterized in that another call terminal receives the call signal and thus makes a call between the call terminals. further,
(S4) The multi-party call method according to claim 9, wherein after the call is finished, a call is made from another call terminal without transmitting right control.

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