JP2007189509A - Call control server, media proxy and congestion control method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To obtain a call control server that can perform congestion control even over a VoIP network which is composed of a plurality of IP terminals connected through a transmission line such as the Internet and uses a connectionless UDP (User Datagram Protocol). <P>SOLUTION: The call control server 7 is equipped with an overload information reception section which receives overload information showing that a transmission line is overloaded, and a control section which controls a transmission section to inhibit a request to establish a new session from being transferred to a second IP terminal even when a connection request reception section receives the request to establish the new session from a first IP terminal 2A to the second IP terminal 2B when the overload information reception section receives the overload information. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to a call control server, a media proxy, and a congestion control method for controlling congestion in a VoIP (Voice over Internet Protocol) network formed by connecting a plurality of IP networks.

  Conventionally, when performing communication on an IP network, there is a limit to the bandwidth of the transmission path, so it has been necessary to manage the data amount of packets flowing through the transmission path in order to ensure communication quality. In addition, when transmitting voice packets using TCP (Transmission Control Protocol), which is a connection-type protocol, information such as MTU (Maximum Transmission Unit) is used to limit the throughput when the bandwidth used increases. Congestion control was performed by this method.

In addition, congestion control in a conventional VoIP network is composed of a public telephone circuit network that is TDM (Time Division Multiplexing) and an IP network as a backbone that integrates these, and VoIP is controlled by controlling the public telephone network. Network congestion control was performed (see, for example, Patent Document 1).
Japanese Patent Laid-Open No. 2002-261925 (FIG. 1)

  However, in a VoIP network, since voice packets are transmitted using connectionless UDP (User Datagram Protocol) instead of connection type TCP, it is difficult to perform congestion control using a method that limits throughput. There was a problem.

  Further, since the congestion control method described in Patent Document 1 is a method for controlling a public telephone network, there is a problem in that it cannot cope with a VoIP network configured by an IP network without using the public telephone network. .

  The present invention has been made to solve the above-described problem, and provides a call control server that enables congestion control even in a VoIP network composed of a plurality of IP terminals connected via a transmission line such as the Internet. To get.

  In the call control server according to the present invention, a receiving unit that receives a session establishment request from the first IP terminal connected in the first station to the second IP terminal connected in the second station; Overload information indicating that a transmission path connecting the transmission unit that receives the session establishment request received by the reception unit to the second IP terminal and the first station and the second station is in an overload state. When the overload information receiving unit and the overload information receiving unit receive the overload information, the connection request receiving unit has received a request for establishing a new session from the first IP terminal to the second IP terminal. Even in this case, the control unit has a control unit that controls the transmission unit so as to prohibit transfer of the session establishment request to the second IP terminal.

  The present invention calculates the transmission rate of this transmission path based on the amount of voice packets passing through the transmission path connecting the first station and the second station, and based on this transmission rate, Since the connection request from the IP terminal to the second IP terminal is controlled, congestion control can be performed even in a VoIP network configured by an IP network.

Embodiment 1 FIG.
FIG. 1 shows a VoIP network according to Embodiment 1 for carrying out the present invention. In FIG. 1, a plurality of IP terminals 2A are connected in the station 1A. Each IP terminal 2A is connected to a hub 3A, and the hub 3A distributes voice packets from each IP terminal 2A to the media proxy 4A and the router 5A. Similarly to the station 1A, the station 1B is connected to a plurality of IP terminals 2B, each IP terminal 2B is connected to a hub 3B, and the hub 3B receives a voice packet from each IP terminal 2B as a media proxy 4B and a router. Distribution to 5B. The router 5A and the router 5B are connected via a transmission line 6, and a call can be made between the IP terminal 2A in the station 1A and the IP terminal 2B in the station 1B. Further, the router 5A and the router 5B are also connected to a call control server 7 that performs call control using SIP (Session Initiation Protocol). SIP is a protocol that creates, converts, and disconnects connections necessary for exchanging voice, video, and text messages to multiple parties in the application layer. In the IETF (Internet Engineering Task Force) Standardization has been performed, and it is defined as a standard track in RFC3261.

  Next, in the case where a VoIP network is configured using SIP as a call control protocol between IP terminals and a call is made between the IP terminal 2A and the IP terminal 2B, FIGS. It explains using. 2 and 3 are block diagrams showing the structures of the media proxy 4 and the call control server 7, respectively, and FIG. 4 is a sequence diagram showing the operation when making a call. As shown in FIG. 2, the media proxy 4A includes a voice packet receiving unit 401 that receives packet data such as voice packets, a voice packet transmitting unit 402 that outputs packet data such as voice packets, and a packet input / output unit 4 that performs input / output. Proxy unit 403 that senses the number of bytes of a packet to be output to transmission line 6, transmission rate calculation part 404 that calculates the transmission rate of transmission line 6, threshold setting part 405 that sets a predetermined threshold, and transmission line 6 is overloaded An overload state determination unit 406 for determining whether the transmission line 6 is overloaded, an overload information generation unit 407 for generating an overload information packet as overload information indicating that the transmission line 6 is in an overload state, and an overload information generation unit The overload information transmission unit 408 transmits the generated overload information. Further, as shown in FIG. 3, the call control server 7 receives a message from the IP terminals 2A and 2B, a reception unit 701, a transmission unit 702 that transmits a message to the IP terminals 2A and 2B, and an overload information packet. An overload information receiving unit 703 and a control unit 704 that controls the transmitting unit 702 are configured. Since the media proxy 4B has the same configuration as the media proxy 4A, a description thereof will be omitted.

  First, when the IP terminal 2A in the station 1A makes a call with the IP terminal 2B in the station 1B, the IP terminal 2A transmits an INVITE message as a connection request to the call control server 7 via the hub 3A and the router 5A. (St101). The call control server 7 transfers the message received by the receiving unit 701 to the IP terminal 2B using the transmitting unit 702 (st102). The transmitting unit 702 transmits a provisional response message 100Trying to the IP terminal 2A (st103). Here, the first three digits (100, etc.) of each message represent a status code as shown in FIG. 5. The first digit is the major classification of the message and the last two digits are the minor classification of the message. Represents. Next, when receiving the INVITE message, IP terminal 2B transmits provisional response 100Trying to call control server 7 (st104). Thereafter, a call process such as ringing a ringing tone of the terminal is performed, and a provisional response 180 Ringing notifying that the call is being performed is transmitted to the call control server 7 (st105). In the call control server 7, the receiving unit 701 receives 180 Ringin, and the transmitting unit 702 transfers this message to the IP terminal 2A (st106).

  In the IP terminal 2B, the connection success message 200OK is transmitted to the IP terminal 2A via the receiving unit 201 and the transmitting unit 202 of the call control server 7 (st107) by an operation such as raising the handset or the like (st108). The IP terminal 2A receives the 200OK message from the IP terminal 2B, and transmits an ACK response (connection establishment acknowledgment) to the IP terminal 2B via the receiving unit 201 and the transmitting unit 202 of the call control server 7 (st109). (St110). With such an operation, a session is established between the IP terminal 2A and the IP terminal 2B, and the connection is completed.

  When the connection is completed, the voice packet from the IP terminal 2A is transmitted to the media proxy 4A via the hub 3A. The media proxy 4A receives the voice packet using the voice packet receiving unit 401, senses the number of bytes of the packet output to the transmission path 6 by this call, and notifies the transmission rate calculation unit 404 as basic information. To do. Thereafter, the voice packet is transferred from the packet transmission unit 402 to the station 1B using the transmission path 6. Note that the proxy unit 403 may perform signal processing such as a voice codec on the voice packet. The station 1B receives this voice packet and the media proxy 4B via the hub 3B, and the media proxy 4B transfers it to the IP terminal 2B. In this way, voice packets are exchanged by UDP on the generated session, and a call state is established (st111).

  Next, the end of the call will be described. When the handset of the IP terminal 2A is placed or the like, a BYE message that is a session disconnection request is transmitted from the IP terminal 2A to the call control server 7 (st112). In the call control server 7, when the BYE message is received by the receiving unit 201, the call control server 7 transfers the BYE message to the IP terminal 2B using the transmitting unit 201 (st113). When receiving the BYE message, IP terminal 2B transmits 200 OK, which is a disconnection acknowledgment, to call control server 7 (st114). The call control server 7 uses the transmission unit 202 to transfer this message to the IP terminal 2A (st115). By such an operation, the session between IP terminal 2A and IP terminal 2B is disconnected, and the call is terminated.

  Next, a congestion control method for the VoIP network configured as described above will be described with reference to FIG. FIG. 6 is a sequence diagram showing an operation at the time of congestion control of the VoIP network.

  First, the same operation as st101 to st111 in FIG. 4 is repeated, and a plurality of IP terminals 2A are in a state where a call is made with the corresponding IP terminal 2B (st201). That is, a plurality of sessions are established, and a call is performed using each session. At this time, the proxy unit 403 senses the number of bytes of the voice packet to be output to the transmission path 6 for each session, and transmits it to the transmission rate calculation unit 404 as basic information. Based on the received basic information, the transmission rate calculation unit 404 calculates the number of transmission bytes per unit time of the established session (hereinafter referred to as the transmission rate), and sends this transmission rate to the overload state determination unit 406. Send.

  A threshold set in advance in the threshold setting unit 405 is transmitted to the overload state determination unit 406, and this threshold is compared with the transmission rate calculated by the transmission rate calculation unit 404. If the transmission rate is larger as a result of the comparison, the overload state determination unit 406 determines that the transmission path 6 is in an overload state (st202), and transmits the determination result to the overload information generation unit 407. In addition, an appropriate value is set as the threshold value from the outside such as an operator. For example, although it is conceivable to set a value of about 80% of the transmission band of the transmission line 6, it is not limited to this value, and an arbitrary value can be set.

  When the overload information generation unit 407 receives the determination result that the transmission line 6 is in an overload state, the overload information generation unit 407 generates an overload information packet including information that the transmission line 6 is in an overload state as overload information. Then, the data is transmitted to the overload information transmission unit 408. The overload information transmission unit 408 transfers this overload information packet to the call control server 7 (st203).

  When the overload information packet is received by the overload information receiving unit 703 from the media proxy 4A, the call control server 7 transfers the overload information packet to the control unit 704. When receiving the overload information packet, the control unit 704 controls the transmission unit 702 to prohibit transfer of a new connection request (st204). In this state, when the call control server 7 transmits an INVITE message as a new connection request from the IP terminal 2A to the IP terminal 2B (st205), the transmission unit 702 sends the request to the IP terminal 2A in the same manner as st103. The provisional response message 100Trying is transmitted (st206), but since it is controlled by the control unit 704, the INVITE message is not transferred to the IP terminal 2B (st207), and the request to the IP terminal 2A is executed. A message 486Busy indicating that it cannot be transmitted is transmitted (st208). For this reason, a new session is not established between the IP terminal 2A and the IP terminal 2B, and congestion control can be performed.

  Further, a case where the currently connected session is disconnected and the transmission rate of the transmission line 6 becomes equal to or less than the threshold will be described. In this case, the transmission rate transmitted to the overload state determination unit 406 is smaller than the threshold value. In addition, the threshold value at this time may be the same value as the threshold value used in st202, or may be a small value. In this way, when setting the threshold value to a different value, it may be set in advance in the threshold value setting unit 404. The overload state determination unit 406 transmits to the overload information generation unit 407 a determination result that the transmission rate is smaller than the threshold and that the overload state is not over. When the overload information generation unit 407 receives a determination result that the transmission path 6 is not in an overload state, the overload information generation unit 407 generates an overload state release packet as overload information and transmits the packet to the overload information transmission unit 408. The overload information transmission unit 408 transfers this overload state release packet to the call control server 7 (st209).

  When the overload state release packet is received by the overload information receiving unit 703 from the media proxy 4A, the call control server 7 transfers the overload information packet to the control unit 704. When the control unit 704 receives the overload state release packet, the control unit 704 releases the control of the transmission unit 702 (st210). Accordingly, when the call control server 7 receives an INVITE message that is a request for establishing a new session for the IP terminal 2B from the IP terminal 2A, the call control server 7 transfers the INVITE message to the IP terminal 2B.

  Also, in the media proxy 4B, since the above-described operations of st202 to st210 are performed, when the transmission rate of the transmission path 6 is larger than the threshold, the connection request is sent from the IP terminal 2B to the IP terminal 2A. Even if there is, a new session can be prevented from being established, and a new session can be established when the transmission rate of the transmission line 6 falls below the threshold.

  In the VoIP network configured in this way, when the transmission rate of the transmission line 6 becomes larger than a preset threshold, it is possible to prevent establishment of a new session, thereby preventing network congestion. It becomes possible to do.

Embodiment 2. FIG.
The VoIP network according to the first embodiment has a configuration using SIP as a call control protocol. It may be 323. H. H.323 is a protocol used for the implementation of so-called videophones, etc., which defines audio, video systems, data compression / decompression systems, etc., for transmitting and receiving audio and video on a network on a one-to-one basis. Approved and recommended by the Union ITU-T.

  Next, as a call control protocol between IP terminals, H.264 is used. A case where a VoIP network is configured using H.323 and a call is made between the IP terminal 2A and the IP terminal 2B will be described with reference to FIG. FIG. FIG. 32 is a sequence diagram illustrating an operation when a call is performed using H.323. Further, the configuration of the VoIP network is the same as that of the first embodiment, and thus the description thereof is omitted.

  First, when the IP terminal 2A in the station 1A makes a call with the IP terminal 2B in the station 1B, a connection request for a session to the IP terminal 2B from the IP terminal 2A to the call control server 7 via the hub 3A and the router 5A. The ARQ, which is a message of a telephone number and connection permission application, is transmitted (st301). When the receiving unit 701 receives this message, the call control server 7 transmits an IP address and a connection permission message ACF to the IP terminal 2A using the transmitting unit 702 (st302). When receiving the message ACF, the IP terminal 2A transmits a message SETUP to set up a call to the IP terminal 2B (st303). Upon receiving this SETUP, IP terminal 2B transmits a telephone number and a connection permission application message ARQ to IP terminal 2A (st304). Upon receiving the message ARQ from the IP terminal 2B, the IP terminal 2A returns an IP address and a connection permission message ACF. The IP terminal 2B that has received this message responds with an ALERT and CONNECT message (st306, st307). When the IP terminal 2A receives the CONNECT message, a session is established between the IP terminal 2A and the IP terminal 2B.

  When the connection is completed, the voice packet from the IP terminal 2A is output to the media proxy 4A via the hub 3A as in the first embodiment. The media proxy 4A receives the voice packet using the voice packet receiving unit 401, senses the number of bytes of the packet output to the transmission path 6 by this call, and notifies the transmission rate calculation unit 404 as basic information. To do. Thereafter, the voice packet is transferred from the voice packet transmitting unit 402 to the station 1B using the transmission path 6. Note that the proxy unit 403 may perform signal processing such as a voice codec on the voice packet. The station 1B receives this voice packet and the media proxy 4B via the hub 3B, and the media proxy 4B transfers it to the IP terminal 2B. In this way, voice packets are exchanged by UDP on the generated session, and a call state is established (st308).

  Next, a congestion control method for the VoIP network configured as described above will be described with reference to FIG. FIG. 8 is a sequence diagram showing an operation at the time of congestion control of the VoIP network.

  First, the same operation as st301 to st308 in FIG. 7 is repeated, and the plurality of IP terminals 2A are in a state where a call is made with the corresponding IP terminal 2B (st401). That is, a plurality of sessions are established, and a call is performed using each session. At this time, as in the first embodiment, the media proxy 4A determines whether or not the transmission line 6 is overloaded based on the transmission rate of the transmission line 6 (st402). If it is determined as an overload state as a result of the determination, an overload information packet is transmitted to the call control server 7 (st403), and an IP address, connection for a new telephone number and connection permission application are transmitted. The transmission unit 702 is controlled to prohibit the return of the permission message ACF (st404). In this state, when a new telephone number and a connection permission application message ARQ message are transmitted from the IP terminal 2A to the call control server 7 (st405), the transmission unit 702 transmits the IP address and the connection permission message. A message indicating that connection is not possible is transmitted without transmitting the ACF (st406). As described above, H. Even if H.323 is used, congestion control can be performed.

  Further, as in the first embodiment, it is needless to say that a new session can be established when the transmission rate of the transmission line 6 falls below the threshold value.

It is a figure which shows the VoIP network in Embodiment 1 and Embodiment 2 for implementing this invention. It is a figure which shows the structure of a media proxy. It is a figure which shows the structure of a call control server. It is a sequence diagram which shows the operation | movement at the time of making a telephone call using SIP It is a figure which shows a status code. It is a sequence diagram which shows the operation | movement at the time of performing congestion control using SIP. H. FIG. 32 is a sequence diagram illustrating an operation when performing a call using H.323. H. 6 is a sequence diagram illustrating an operation when performing congestion control using H.323. FIG.

Explanation of symbols

1A, 1B Station 2A, 2B IP terminal 4A, 4B Media proxy 401 Voice packet receiver 402 Voice packet transmitter 404 Transmission rate calculator 406 Overload state determiner 408 Overload information transmitter 6 Transmission path 7 Call control server 701 Reception Unit 702 transmitting unit 703 overload information receiving unit 704 control unit

Claims (5)

A receiving unit that receives a session establishment request from the first IP terminal connected in the first station to the second IP terminal connected in the second station;
A transmission unit that transfers a session establishment request received by the reception unit to the second IP terminal;
An overload information receiving unit for receiving overload information indicating that a transmission line connecting the first station and the second station is in an overload state;
When the overload information receiving unit receives the overload information, even if the connection request receiving unit receives a request for establishing a new session from the first IP terminal to the second IP terminal, A control unit that controls the transmission unit to prohibit transfer of an establishment request to the second IP terminal;
A call control server comprising: A receiver for receiving a session establishment request from the first IP terminal connected in the first station to the second IP terminal connected in the second station;
A transmitter for returning a message for establishing the session to the first IP terminal;
An overload information receiving unit for receiving overload information indicating that a transmission line connecting the first station and the second station is in an overload state;
When the overload information receiving unit receives the overload information, even if the connection request receiving unit receives a request for establishing a new session from the first IP terminal to the second IP terminal, A control unit that controls the transmission unit to prohibit a reply to the first IP terminal for establishing a message;
A call control server comprising: A voice packet receiver for receiving voice packets from a first IP terminal connected in the first station;
A voice packet for transferring a voice packet received by the voice packet receiving unit to a second IP terminal connected in a second station via a transmission path connecting the first station and the second station. A transmission unit;
A transmission rate calculation unit for calculating a transmission rate of the transmission line based on the amount of voice packets transmitted through the transmission line;
Based on the transmission rate calculated by the transmission rate calculation unit, an overload state determination unit that determines whether the transmission line is in an overload state,
When the overload state determination unit determines that the transmission path is in an overload state, the call control server that performs call control between the first IP terminal and the second IP terminal, An overload information transmission unit that transmits overload information indicating that the transmission line is in an overload state;
A media proxy, comprising: 3. The media proxy according to claim 2, wherein the overload state determination unit determines that the transmission path is in an overload state when the transmission rate is greater than a predetermined threshold. A first IP terminal connected within the first station;
A second IP terminal connected in the second station;
A transmission line connecting the first station and the second station;
A call control server that performs call control between the first IP terminal and the second IP terminal;
A media proxy that receives a voice packet output from the first IP terminal and transfers the voice packet to the second IP terminal using the transmission path;
A congestion control method for controlling congestion of a VoIP network having:
A transmission rate calculating step of calculating a transmission rate of the transmission line based on a voice packet transmitted through the transmission line;
Based on the transmission rate calculated in the transmission rate calculation step, an overload state determination step for determining whether or not the transmission line is in an overload state;
When it is determined in the overload state determination step that the transmission path is overloaded, overload information indicating that the transmission path is overloaded is transmitted to the call control server. An overload information transmission step;
When the call control server receives the overload information, the call control server prohibits transmission of a message for establishing the session even when the session establishment request from the first IP terminal to the second IP terminal is received. A control step for controlling to
The congestion control method characterized by having.

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