JP2009141939A - Communication system and codec allocation method thereof - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a communication system and codec allocation method thereof in which, a codec of data to be transmitted is varied according to a status of an IP network, by automatically allocating the codec of the data to be transmitted in accordance with a bandwidth available for the network when performing communication while utilizing the network. <P>SOLUTION: A communication system of the present invention includes: a communication control section which is connected with a plurality of terminals each including a speech function utilizing an IP network, for processing a call in response to a speech connection requirement from a first terminal; a bandwidth management section for measuring the amount of bandwidths used in a first network to which the first terminal, that has required a speech connection, is connected, in response to the speech connection requirement; and a codec allocation section for allocating an audio codec or a video codec to the first terminal, that has required the speech connection, in accordance with a measurement result of the bandwidth management section, during call processing due to the communication control section. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  According to the present invention, when communication is performed using an IP network, the codec of data to be transmitted is automatically assigned in accordance with the available bandwidth of the network, so that the data is transmitted according to the state of the network. The present invention relates to a communication system and a method for assigning the codec.

  The present invention is derived from research conducted as part of leading technology test support of the Information Communication Department and the Information Communication Research Promotion Field [Problem Management Number: 2007-Base-05, Issue Name: Next Generation Network Test Technology support].

  In general, VoIP (Voice over Internet Protocol) uses a data communication packet network that has been used in the past for calls over the Internet, and transmits voice data converted into Internet protocol data packets. Is a communication service technology that enables communication with a general telephone network.

  Such a VoIP service can be used when a computer, an IP telephone, or a dedicated router is provided, so that it is convenient to use, and a plurality of VoIP services can be used simultaneously via a cable, and has excellent extensibility. The amount of usage is gradually increasing because it is much cheaper than existing phones.

  In the VoIP service as described above, when communication is performed via the Internet using a terminal such as an IP telephone or a computer, communication is performed with the audio and video codecs of the subscriber terminal being always fixed. Thereby, even when the subscriber network has a large available network bandwidth, it provides low quality using low-quality audio and video codecs, or when the subscriber network has no available network bandwidth, There is a problem in that the quality for the VoIP flow cannot be provided in the IP network using the high bandwidth audio and video codec.

  In addition, since the BE (Best Effort) communication method is used in the case of a VoIP telephone service using the Internet, there are many simultaneous subscribers by applying a method in which a large number of subscribers share the entire bandwidth. However, there is a problem that the transmission speed is lowered for all connected subscribers.

  The present invention has been made in view of the above-described problems, and an object of the present invention is to automatically set a data codec corresponding to the available bandwidth of a network when transmitting data including voice or video in communication using an IP network. It is an object of the present invention to provide a communication system and a codec allocation method for increasing the number of simultaneous users and improving the quality of service.

  In order to achieve the above object, a communication system according to the present invention is connected to a plurality of terminals including a call function using an IP network, and processes a call in response to a call connection request of a first terminal; In response to a call connection request of the first terminal, a bandwidth management unit that measures the bandwidth usage of the first network connected to the first terminal, and the bandwidth during call processing of the communication control unit Corresponding to a measurement result of the management unit, a codec allocating unit that allocates an audio codec or a video codec to the first terminal that has requested a call connection.

  The codec allocation method in the communication system according to the present invention includes a step of requesting a call connection from a first terminal of a first network to which a plurality of terminals are connected to a second terminal, and a bandwidth of the first network. Measuring a usage amount, and automatically assigning a codec of audio or video data to the first terminal according to the usage amount of the bandwidth; setting a call to the first terminal; Voice and video data are transmitted from the first terminal according to a codec, and a call with the second terminal is performed.

  According to the present invention, a communication system and a codec assignment method thereof are capable of efficiently transmitting data by automatically assigning a codec corresponding to the available bandwidth of the network, and setting a dynamic codec. It is possible to provide high quality audio and video to connected subscribers when bandwidth usage is low, and when there are many simultaneous users and high bandwidth usage. By maintaining a certain level of quality while accommodating connected users by changing the codec, there is an effect that the quality of service quality management and the management efficiency with respect to the number of simultaneous connected users are improved.

  Hereinafter, embodiments of the present invention will be described with reference to the accompanying drawings.

  FIG. 1 is a diagram which is referred to for a description of a configuration of a communication system using an IP network according to an embodiment of the present invention.

  As shown in FIG. 1, a communication system according to the present invention allocates call settings and codecs to a plurality of terminals connected to a plurality of terminals 111 to 116 having a call function via an IP network. Communication servers 210 and 220 and gateways 310 and 320 that allow a plurality of terminals to be connected to the IP network 400 are provided.

  The communication system further includes a codec conversion device 410 that converts the codec of data transmitted in response to a request from the communication servers 210 and 220. At this time, the codec conversion apparatus 410 may be provided in the IP network 400 or connected to the IP network 400.

  The communication system configured as described above converts the audio signal and the video signal into compressed data packets and transmits them through the Internet based on the IP network. Communication is performed in conjunction with a certain PSTN network.

  At this time, the terminal is a communication VoIP (Voice over Internet Protocol) service dedicated IP telephone via the IP protocol, a computer (PC) that provides an inter-communication voice service having a communication ID or IP address via the Internet, A general telephone provided with a dedicated router can be used, and a PDA or a mobile phone capable of IP-based communication can also be used. The terminal as described above includes audio input means and video input means, and transmits and receives packets and data processing means for processing the input audio or video according to the codec assigned by the communication servers 210 and 220. Packet communication means may be provided.

  The terminal as described above is a PC-to-PC that provides an inter-communication voice service having a communication ID, IP address, or the like via the Internet when the communication is performed via the Internet. The IP protocol for interworking the Internet and the PSTN network and the IP telephone-to-general telephone form for providing voice service between the incoming call or the general wired / wireless telephone, and the PSTN network general telephone is connected to another PSTN network via the Internet. Communication can be performed in any one form of a general telephone to a general telephone connected to the general telephone.

  The present invention can also be applied to communication by interlocking with a general telephone network, but will be described by taking the case of IP telephone to IP telephone, PC to PC or IP telephone to PC as an example. Hereinafter, description of connection with the general telephone network is omitted.

  The communication servers 210 and 220 are connected to a plurality of terminals 111 to 116 and set and control a VoIP call according to a request from the terminals so that communication between the terminals using the IP network 400 is possible. At this time, the first communication server 210 controls the communication to be performed according to the request of the connected first to third terminals 111 to 113, and the second communication server 220 controls the fourth to sixth terminals 114 to 116. Control communication.

  In addition, the communication servers 210 and 220 manage the usable bandwidth of the network connected to the institutional subscriber section, that is, the connected terminal, and measure the bandwidth usage of the network in real time. At this time, the communication servers 210 and 220 generate a bandwidth management table for bandwidth usage.

  The communication servers 210 and 220 refer to the stored bandwidth management table and assign codecs to the terminals according to the measured bandwidth usage. Here, the bandwidth management table is set so that the amount of bandwidth used is divided into predetermined stages, and different codecs are allocated to terminals according to the respective stages. Accordingly, the terminal transmits audio or video data according to the codec assigned from the communication servers 210 and 220, and communicates with the other party.

  Further, when a call is received from another terminal, the communication servers 210 and 220 request the codec conversion apparatus 410 to perform codec conversion by comparing the codec for the received call with the codec assigned to the terminal.

  For example, when a call is received from the fourth terminal 114 connected to the second communication server 420 to the second terminal 112 connected to the first communication server 410, the first communication server 410 responds to the bandwidth usage. When the codec is assigned to the second terminal 112 and the codec of the data for the call reception of the fourth terminal 114 is different from the codec assigned to the second terminal 112, the codec conversion device of the IP network 400 A codec conversion is requested to 410.

  At this time, the codec conversion apparatus 410 re-encodes the codec of the audio or video data transmitted in response to the request from the communication servers 210 and 220 to the codec requested from the communication servers 210 and 220, and converts the codec. . That is, the codec converter 410 re-encodes and converts the codec of the data transmitted from the fourth terminal 114 to the codec assigned to the second terminal 112 in response to a request from the first communication server 210. The transmitted data is transmitted to the first communication server 210.

  FIG. 2 is a block diagram referred to in the description of the configuration of the communication server of the communication system according to the embodiment of the present invention.

  As illustrated in FIG. 2, the communication server 200 includes a call control unit 211, a bandwidth management unit 214, a codec allocation unit 212, a data unit 213, and a communication unit 215.

  The call control unit 211 sets a VoIP call to the connected terminal in response to a request from the terminal, and establishes a call path with the corresponding terminal when a call is received from another terminal to enable mutual communication. The data unit 213 stores a bandwidth management table generated by the bandwidth management unit 214 according to the amount of bandwidth used, and the communication unit 215 transmits / receives data to / from a plurality of connected terminals, via a gateway. Communication is performed via the IP network so that control signals are transmitted and received.

  The bandwidth management unit 214 measures the bandwidth usage of the network in real time and manages the bandwidth. In addition, the bandwidth management unit 214 generates a bandwidth management table according to the measured bandwidth usage.

The bandwidth management table generated by the bandwidth management unit 214 is as shown in Table 1 below.
At this time, the bandwidth management table is a table in which the amount of bandwidth used is divided into predetermined stages, and the codecs to be used for each stage are arranged according to the available bandwidth. The higher the available bandwidth, the higher the quality audio and video codec is used.

  As shown in Table 1, the bandwidth management table divides the bandwidth usage into 4 levels: 30% or less, 30% excess 40% or less, 40% excess 50% or less, 50% or more. G. 711, G.G. 726, G.G. 728, G.G. 729, G.G. 723 audio codec; H.264, H.H. 263, H.M. The codec is designated so that the video codec of 261 is assigned according to the quality.

  At this time, when the data link is Ethernet (registered trademark), the bandwidth of the voice codec is G.264. 711 is 87.2 Kbps. 726 is 55.2 Kbps to 27.6 Kbps. 728 is 27.6 Kbps to 39.2 Kbps, G. 729 is 19.6 Kbps to 31.2 Kbps. 723 is between 12.9 Kbps and 21.5 Kbps.

  In the bandwidth management table as described above, the stage classification according to the bandwidth usage and the type of codec to be assigned can be changed according to the bandwidth of the institutional subscriber and the policy of the administrator. In addition to the designated codec, a table can be set by adding a codec according to the policy and technical development of the administrator, and according to circumstances, the table can be configured with only the audio codec excluding the video codec.

  The codec allocation unit 212 allocates a codec to the terminal that has requested the call setting in accordance with the bandwidth usage measured by the bandwidth management unit 214.

  The codec allocation unit 212 allocates the highest quality audio codec and video codec to the terminal when the bandwidth usage of the network is 30% or less according to the codec management table as described above, and the bandwidth usage is 30. If the percentage exceeds 40% or less, an intermediate quality audio codec and video codec are allocated to the terminal, and if the bandwidth usage exceeds 40% and 50% or less, the lowest quality audio codec and video codec are allocated. In addition, when the bandwidth usage exceeds 50%, the codec allocation unit 212 allocates the lowest quality audio codec to the terminal except for the video codec so that the video codec does not operate.

  For example, as shown in Table 1, the codec allocation unit 212 has the highest quality speech codec G.D when the bandwidth usage is 30% or less. 711 and video codec H.264. H.264 is assigned to the terminal, and the terminal starts communication with the designated codec.

  Further, the codec allocating unit 212 is also received according to the bandwidth usage measured by the bandwidth management unit 214 even when a call is received by any one of the connected terminals. Assign a codec to the target terminal.

  At this time, the codec allocation unit 212 compares the codec of the data for the received call with the codec allocated to the target terminal, and requests the codec conversion device 410 of the IP network 400 for codec conversion when the codecs are different from each other. To do.

  The operation of the embodiment of the present invention configured as described above will be described as follows.

  FIG. 3 is a flowchart referred to in the description of the operation for the codec allocation method corresponding to the data transmission in the communication system according to the embodiment of the present invention.

  As shown in FIG. 3, when a call connection request is received from the connected terminal 100 (S510), the communication server 200 measures the current network bandwidth usage (S520).

  The communication server 200 sets a VoIP call to the terminal that requested the call connection, and refers to the bandwidth management table in which the audio and video codec for each stage corresponding to the bandwidth usage is specified, and the bandwidth usage A codec corresponding to is assigned to the terminal.

  At this time, the communication server 200 compares the plurality of reference values with the corresponding audio codec or video codec according to the bandwidth usage level set in the bandwidth management table. The lower the allocation and bandwidth usage, that is, the higher the available bandwidth, the higher the quality codec is allocated to the terminal.

  When the bandwidth usage is equal to or smaller than the first reference value (S530), the communication server 200 allocates a high-quality first audio codec and first video codec corresponding to the large available bandwidth. (S540).

  If the bandwidth usage is equal to or greater than the first reference value excess second reference value (S550), the second audio codec and the third video codec are allocated to the terminal (S560), and the bandwidth usage is If the second reference value exceeds the third reference value (S570), the third audio codec and the third video codec are allocated to the terminal (S580).

  On the other hand, if the bandwidth usage exceeds the third reference value (S590), only the fourth audio codec is allocated to the terminal without assigning the video codec (S610). At this time, if the bandwidth usage is not included in any stage, it is determined that an error has occurred (S600).

  For example, as shown in Table 1 above, when the bandwidth usage is 30% or less, G. 711 or G.I. 726 audio codec; H.264 or H.264 If the video codec of H.263 is allocated to the terminal and the bandwidth usage is more than 30% and 40% or less, G. 726 or G.I. 728 voice codec and H.264. H.264 or H.264 263 video codecs are allocated to the terminal. Further, when the bandwidth usage is more than 40% and 50% or less, G.I. 729 voice codec and H.264. If the video codec of H.261 is allocated and the bandwidth usage exceeds 50%, the G. The audio codec of 723 is assigned to the terminal so that only audio data packets are transmitted.

  As described above, when an audio codec or a video codec is allocated to a terminal by the communication server 200 and a call is connected, the terminal 100 starts communication by transmitting audio or video data according to the allocated codec. (S620).

  FIG. 4 is a flowchart referred to in the operation description for the codec assignment and change method according to the data reception in the communication system according to the embodiment of the present invention.

  As shown in FIG. 4, the communication server 200 receives codec information for a received call together with a call connection request when a call is received by any one of a plurality of connected terminals (S710). (S720).

  That is. When a call is received from the fourth terminal 114 connected to the second communication server 220 to the first terminal 111 connected to the first communication server 210 via the IP network, the first communication server 210 receives the call from the fourth terminal 114. , The codec of the data of the fourth terminal 114 is confirmed.

  The first communication server 210 measures the bandwidth usage of the network to which the first terminal 111 is connected to confirm the available bandwidth (S730), and sends the audio codec to the first terminal 111 according to the bandwidth usage. Alternatively, a video codec is assigned (S740 to S800).

  At this time, as shown in FIG. 3 described above, codec allocation is performed by dividing a bandwidth usage amount into predetermined stages, and a predetermined audio codec or video designated according to a comparison result with each stage reference value. A codec is assigned to the first terminal 111. When the bandwidth usage exceeds the third reference value, that is, when the bandwidth usage exceeds 50%, only the fourth voice codec is allocated to the first terminal 111 as described above ( S800).

  The first communication server 210 compares the codec assigned to the first terminal 111 and the codec for the received call of the fourth terminal 114 as described above (S810), and if both codecs are different from each other, the codec conversion is performed. The device 410 is requested to perform codec conversion (S820).

  At this time, the codec conversion apparatus 410 receives the codec of the audio or video data transmitted from the fourth terminal 114 to the first terminal 111 in response to a request from the first communication server 210, and is assigned to the first terminal 111. Alternatively, it is changed to a video codec, and the codec converted audio or video data is transmitted to the first communication server 210.

  The first communication server 210 establishes a call path in response to a call connection request from the fourth terminal 114, connects the call to the first terminal 111, and starts a mutual call (S830).

  Therefore, it is possible to use the highest quality codec when the available bandwidth is large by variably applying the codec for audio or video data corresponding to the bandwidth usage, thus preventing degradation of service quality Even when there are many simultaneous users and the available bandwidth is small, it is possible to accommodate the users by changing the codec.

  The above-described preferred embodiments of the present invention have been disclosed for the purpose of illustration, and those having ordinary knowledge in the technical field to which the present invention pertains depart from the technical idea of the present invention. Various substitutions, modifications, and alterations are possible within the scope of not being included, and such substitutions, alterations, and the like belong to the scope of the claims.

It is a figure referred by the description with respect to the structure of the communication system using the IP network by embodiment of this invention. It is a block diagram referred by the description with respect to the structure of the communication server of the communication system by embodiment of this invention. 5 is a flowchart referred to in an operation description for a codec allocation method according to data transmission in a communication system according to an embodiment of the present invention. 5 is a flowchart referred to in an operation description for a codec assignment and change method according to data reception in a communication system according to an exemplary embodiment of the present invention;

Claims (12)

A call connection request from the first terminal of the first network to which the plurality of terminals are connected to the second terminal;
Measuring bandwidth usage of the first network and automatically assigning codec of audio or video data to the first terminal corresponding to the bandwidth usage;
Communicating with the second terminal by setting up a call to the first terminal and transmitting audio and video data from the first terminal according to the assigned codec. Codec assignment method in the system. When a call is received from the second terminal to the first terminal, automatically assigning a codec corresponding to the bandwidth usage of the first network connected to the first terminal;
When the codec of the data of the second terminal and the codec assigned to the first terminal are compared, and the codec of the data of the second terminal and the codec assigned to the first terminal are different from each other, The method for assigning codecs in a communication system according to claim 1, further comprising: requesting codec conversion.   The codec allocating step divides bandwidth usage into predetermined stages, and assigns high-quality audio and video codecs in stages when the bandwidth usage is low, and uses the bandwidth usage. The codec assignment method in the communication system according to claim 1 or 2, wherein when there are many, a low-quality audio codec or video codec is assigned step by step.   The codec allocation step refers to a bandwidth management table in which audio codecs or video codecs having different qualities are specified for each stage, and assigns the audio codec or video codec corresponding to the bandwidth usage to the first terminal. The method for assigning codecs in a communication system according to claim 3, wherein the codec is assigned to the codec.   The codec assignment step further includes a step of assigning only an audio codec excluding a video codec when the bandwidth usage is equal to or greater than a predetermined reference value. Assignment method.   3. The codec allocation method in a communication system according to claim 2, further comprising a step of converting a codec of data of the second terminal into a codec of the first terminal and transmitting the codec to the first terminal. A communication control unit that is connected to a plurality of terminals including a call function using an IP network and processes a call in response to a call connection request of the first terminal;
In response to the call connection request of the first terminal, a bandwidth management unit that measures the bandwidth usage of the first network connected to the first terminal;
A codec allocating unit that allocates an audio codec or a video codec to the first terminal in response to a measurement result of the bandwidth management unit during call processing of the communication control unit.   The codec allocating unit automatically refers to an already stored bandwidth management table and automatically assigns an audio codec or a video codec to the first terminal according to a bandwidth usage input from the bandwidth management unit. The communication system according to claim 7 to be assigned.   9. The communication system according to claim 8, wherein the codec allocating unit allocates only an audio codec to the first terminal except for a video codec when the bandwidth usage is equal to or greater than a predetermined reference value. .   The codec allocating unit allocates a codec to the first terminal to be received when a call is received from a second terminal to the first terminal among the plurality of terminals, and a codec for the received call of the second terminal and the 8. The codec conversion device is requested to perform codec conversion so that the codec of the received call is converted into the codec allocated to the first terminal when the codec allocated to the first terminal is different from each other. Communication system.   The codec conversion apparatus is connected to the IP network and re-encodes data transmitted or received to the first terminal in response to a request from the codec allocating unit to a requested codec and transmits the data. The communication system described.   9. The bandwidth management unit according to claim 8, wherein the bandwidth management unit divides the bandwidth usage into predetermined stages, sets different codecs for each stage, and generates the bandwidth management table. Communication system.

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