Abstract:
A method of changing a fragment size of data packets in a router where a data packet is divided into the data packets having the fragment size, and are transmitted to a network along with audio packets includes the steps of acquiring, in the router, a parameter indicative of whether proper audio quality is maintained through transmission of the audio packets, and changing the fragment size of the data packets in response to the acquired parameter.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention generally relates to a router and a method of changing a fragment size of data packets, and particularly relates to a router connected to a network conveying data packets and audio packets and a method of changing a fragment size of data packets that are supplied to the network. 
   2. Description of the Related Art 
     FIG. 1  is an illustrative drawing for explaining a VoIP (voice over internet protocol) router. 
   As shown in  FIG. 1 , a VoIP router  11  is provided between a WAN (wide are network), a LAN (local area network), and a PBX (private branch exchange)  10 . The VoIP router  11  converts data signals and audio signals into packets when the data signals are supplied from the LAN, and the audio signals are supplied from the PBX  10 , and sends the packets to the WAN. When receiving data packets and audio packets from the WAN, the VoIP router converts these packets into data signals and audio signals, which are then supplied to the LAN and the PBX  10 , respectively. 
   The VoIP router  11  establishes interface with the LAN, the WAN and the PBX  10 . 
   In the VoIP router  11 , there is a need to avoid making an audio frame wait until transmission of a packet to the WAN is finished where the packet may be such a long packet as that of FTP (file transfer protocol) or HTTP (hypertext transport protocol). To this end, such a long packet is divided, and audio packets are inserted therebetween. This is called fragmentation. The VoIP router checks an MTU (maximum transfer unit) size of the IP (Internet protocol) layer. When the router receives a packet having a size exceeding the MTU size, the router notifies, via ICMP (Internet control message protocol), the source of the packet that the excess size of the packet creates errors, and notifies the source how large the MTU size is. An apparatus at the packet source adjusts the packet size to the MTU size, and transmits packets having a shorter size. 
   In this configuration, when the VoIP router receives a packet having a size exceeding the MTU size, the VoIP router arranges for the source to transmit shorter packets matching the MTU size. Alternatively, the VoIP router may change the packet to a shorter packet that conforms to the MTU size. 
   Data that is transmitted via FTP or HTTP forms a packet as large as 1000 bytes, for example. Audio packets, on the other hand, have a size that is as small as a two-digit figure in byte. 
     FIG. 2  is an illustrative drawing for explaining transmission of data from a router. 
   As shown in  FIG. 2 , a long packet D may be divided into shorter packets D 1  through D 5 , which are then transmitted to the WAN while audio packets V 1  through V 4  having priority are inserted between the shorter packets D 1  through D 5 . Even in this case, transmission of the audio packets may be delayed if the MTU size is relatively large, thereby degrading audio quality. 
   For example, if the MTU size is so large that the data packets D 1  through D 5  are significantly larger than the audio packets V 1  through V 5 , the audio packets V 1  through V 3  are delayed by the data packet D 5 , and the audio packet V 1  is further delayed by the data packet D 4 . 
   In general, the shorter the fragment size of data, the higher the audio quality is. However, improvement in the audio quality is achieved at the expense of the throughput of data communication. Accordingly, if sufficient audio quality is being maintained, the fragment size of data may be lengthened to boost the throughput of data communication. 
   Conventionally, the MTU size is fixed, and does not change dynamically to cope with situational changes. 
   Accordingly, there is a need for a scheme that can automatically change a fragment size of a data packet so as to keep audio quality within a predetermined range. 
   SUMMARY OF THE INVENTION 
   It is a general object of the present invention to provide a router and a method of changing a fragment size that substantially obviate one or more of the problems caused by the limitations and disadvantages of the related art. 
   Features and advantages of the present invention will be set forth in the description which follows, and in part will become apparent from the description and the accompanying drawings, or may be learned by practice of the invention according to the teachings provided in the description. Objects as well as other features and advantages of the present invention will be realized and attained by a router and a method particularly pointed out in the specification in such full, clear, concise, and exact terms as to enable a person having ordinary skill in the art to practice the invention. 
   To achieve these and other advantages and in accordance with the purpose of the invention, as embodied and broadly described herein, the invention provides a method of changing a fragment size of data packets in a router where a data packet is divided into the data packets having the fragment size, and are transmitted to a network along with audio packets, including the steps of acquiring, in the router, a parameter indicative of whether proper audio quality is maintained through transmission of the audio packets, and changing the fragment size of the data packets in response to the acquired parameter. 
   In the method described above, the parameter that indicates whether proper audio quality is maintained is acquired, and is consulted to change the fragment size of the data packets. This makes it possible to improve data throughput while securing proper audio quality. 
   According to the present invention, the parameter is selected from a wait time of the audio packets, a delay time of the network, the number of congestion notices, and the number of audio calls. The wait time is a time period for which the audio packets wait in the router before being transmitted to the network. The delay time of the network is a time period that passes from transmission of a hello packet to reception of the hello packet returning from the network. The number of congestion notices indicates how many times a congestion notice is received from the network during a predetermined time period. The number of audio calls indicates the number of audio calls simultaneously taking place in the router. Use of one of these parameters makes it possible to improve data throughput while securing proper audio quality. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is an illustrative drawing for explaining a VoIP router; 
       FIG. 2  is an illustrative drawing for explaining transmission of data from a router; 
       FIG. 3  is an illustrative drawing showing a system to which the present invention is applied; 
       FIG. 4  is a block diagram showing configurations of VoIP routers and a gatekeeper of  FIG. 3 ; 
       FIGS. 5A and 5B  are tables showing data structures of a gatekeeper table and a routing table, respectively; 
       FIG. 6  is a block diagram of the VoIP router; 
       FIG. 7  is a flowchart of a first method of adjusting a fragment size; 
       FIG. 8  is an illustrative drawing for explaining how to determine a fragment size based on a wait-time deviation; 
       FIG. 9  is a flowchart of a second method of adjusting a fragment size; 
       FIG. 10  is a flowchart of a third method of adjusting a fragment size; 
       FIG. 11  is a flowchart of a fourth method of adjusting a fragment size; and 
       FIG. 12  is a flowchart of a fifth method of adjusting a fragment size. 
   

   DESCRIPTION OF THE PREFERRED EMBODIMENTS 
   In the following, embodiments of the present invention will be described with reference to the accompanying drawings. 
     FIG. 3  is an illustrative drawing showing a system to which the present invention is applied. 
   The system of  FIG. 3  is made up from points A through E that are connected to a WAN  100 . In  FIG. 3 , points A, B, C, D, and E are additionally referenced by reference numerals  20 A,  20 B,  20 C,  20 D and  20 E, respectively. The WAN  100  is comprised of dedicated lines, frame-relay networks, ATM networks, and the like. The point A is comprised of a PBX  21 A, VoIP router  22 A, a server  23 , and a gatekeeper  24 . The points B through E have an identical configuration, and include PBXs  21 B through  21 E, VoIP routers  22 B through  22 E, and personal computers  25 B through  25 E, respectively. 
   The VoIP routers  22 A through  22 E are connected to each other via the WAN  100 . The point A plays a key role in the system of  FIG. 3 , and attends to inter-computer communication (e.g., between the server  23  of the point A and the personal computer  25 B of the point B) as well as inter-PBX audio communication (e.g., between the PBX  21 A of the point A and the PBX  21 B of the point B via the VoIP router  22 A of the point A). The WAN conveys both the data packets and the audio packets. 
     FIG. 4  is a block diagram showing configurations of the VoIP routers and the gatekeeper. 
   The VoIP router  22  converts data signals and audio signals into IP frames, and transmits the IP frames. In  FIG. 4 , any one of the VoIP routers  22 A through  22 C includes a control unit  30 , a routing table  31 , a WAN-interface unit  32 , a routing unit  33 , an audio-interface unit  34 , and a LAN-interface unit  35 . The LAN-interface unit  35  is connected to the server  23  or the personal computer  25 B or  25 C via a LAN. The audio-interface unit  34  is connected to the PBX  21 A,  21 B, or  21 C. 
   The control unit  30  of the VoIP router attends to overall control of the VoIP router. In detail, the control unit  30  arranges for the LAN-interface unit  35  to attend to packet-dividing/assembling operation, and arranges for the routing unit  33  to attend to packet-priority-control operation. Further, the control unit  30  updates the routing table  31  as it becomes necessary through communication with other VoIP routers, and conducts communication with the gatekeeper  24 . 
   The LAN-interface unit  35  establishes interface with a LAN such as 10/100BASE. In detail, the LAN-interface unit  35  divides a long packet, and assembles divided packets under the control of the control unit  30 . 
   The audio-interface unit  34  establishes interface with the PBX  21 A,  21 B, or  21 C. In detail, the audio-interface unit  34  digitizes audio signals and signaling signals, and hands the digitized signals to the routing unit  33 . Further, the audio-interface unit  34  detects signaling information (e.g., call-transmission information, call-reception information, phone-number information, and so on), and informs the control unit  30 . 
   The routing unit  33  delivers received packets to the WAN-interface unit  32 , the audio-interface unit  34 , and the LAN-interface unit  35  according to their destinations. Selection of an interface unit is made by referring to a routing table by using an address portion of a packet header. Further, the routing unit  33  has a queue for packet transmission, and adjusts a transmission order and transmission timings under the control of the control unit  30 . 
   The routing table  31  is a table that stores correspondences between IP addresses and VoIP routers. 
     FIGS. 5A and 5B  are tables showing data structures of a gatekeeper table and a routing table, respectively. 
   As shown in  FIG. 5B , the VoIP routing table includes network addresses, costs, and relay routers. 
   In the routing table  31  of the VoIP router  22 A at the point A, for example, the address 127.0.1.1 is listed together with cost “0” and no relay router. This address 127.0.1.1 indicates the address of the server  23  provided at the point A. The cost of the server  23  indicates the number of intervening routers from the VoIP router  22 A to the server  23 , and, thus, is zero in this case. Since there is no need for relaying, no entry is given in the field for the relay router. 
   Further, the address 127.0.3.1 at the point A is listed together with cost “1” and two relay routers having addresses 128.0.3.1 and 129.0.3.1. 
   The routing table  31  of the point B and the routing table  31  of the point C are structured in the same manner as the routing table  31  of the point A. 
   The gatekeeper  24  includes a control unit  40 , a gatekeeper table  41 , a LAN-interface unit  42 , and an address-notifying unit  43 . 
   The control unit  40  of the gatekeeper  24  attends to overall control of the gatekeeper. In details, the control unit  40  detects current audio communication conditions, and updates the gatekeeper table. 
   The gatekeeper table  41  is a table in which phone numbers are stored with matching IP addresses. Communication flags are also stored for the purpose of management and control of audio communication conditions. 
   As shown in  FIG. 5A , the gatekeeper table  41  includes prefix numbers, VoIP-router addresses, and communication flags. 
   As shown in  FIG. 5A , the VoIP-router address of a PBX at the point A having the prefix number 7000 is 127.0.2.1. The VoIP-router address of a PBX at the point B having the prefix number 7001 is 128.0.2.1. The VoIP-router address of a PBX at the point C having the prefix number 7002 is 129.0.2.1. 
   The gatekeeper table  41  is used for controlling the prefix numbers. On the other hand, extension numbers are controlled by the PBX. A communication flag that is 1 indicates an ongoing status of communication, and a communication flag that is 0 indicates no current communication. 
   The LAN interface unit  42  establishes interface with a LAN such as 10/100BASE. 
   The address-notifying unit  43  refers to the gatekeeper table  41 , and responds to an inquiry of a phone number or an IP address when it is issued from a VoIP router. 
   Operation of the configuration of  FIG. 4  will be described below with reference to an example in which communication is simultaneously conducted between the point A and point B and between the point A and the point C. 
   [Telephone Communication Between A and B] 
   A phone call is made from a phone connected to the PBX  21 A at the point A to a phone connected to the PBX  21 B at the point B. A procedure for establishing this communication will be described below. 
   1. When a call is made from the phone connected to the PBX  21 A of the point A to the phone at 7001-xxxx that is connected to the PBX  21 B of the point B, the PBX  21 A at the point A ascertains from the prefix of the call that the call is not directed to itself but directed to an outside station. The PBX  21 A sends signaling information to the VoIP router  22 A. 
   2. The audio-interface unit  34  of the VoIP router  22 A forwards the signaling information to the control unit  30 , and digitizes it. 
   3. The control unit  30  sends an inquiry to the gatekeeper  24  to learn an IP address of the VoIP router corresponding to the prefix number 7001. 
   4. The address-notifying unit  43  of the gatekeeper  24  refers to the gatekeeper table  41  to obtain the IP address 128.0.2.1 of the audio-interface unit  34  of the VoIP router  22 B corresponding to the prefix number 7001, and sends the obtained IP address to the VoIP router  22 A as a reply to the inquiry. Further, the control unit  40  of the gatekeeper  24  detects a start of audio communication between the VoIP router  22 A and the VoIP router  22 B, and sets a communication flag in the relevant table. 
   5. The control unit  30  of the VoIP router  22 A sends the received IP address to the routing unit  33  when the IP address is received from the gatekeeper  24 . The routing unit  33  at the point A consults the routing table  31 , and finds an IP address 127.0.3.1 as an address to which the call is directed. Then, a packet directed to the VoIP router  22 B is generated, and is send to the WAN-interface unit  32  of the point A. 
   6. The WAN-interface unit  32  at the point A transmits the packet to the WAN  100 . 
   7. The WAN-interface unit  32  at the point B receives the packet from the VoIP router  22 A, and passes the packet to the routing unit  33 . 
   8. The routing unit  33  at the point B refers to the routing table  31  at the point B, and ascertains that the packet is directed to the audio-interface unit  34  of the point B. The packet is then sent to the audio-interface unit  34  of the point B. 
   9. The audio-interface unit  34  at the point B disassembles the packet, and converts the signaling information into an analog signal, which is then sent to the PBX  21 B. 
   10. The PBX  21 B makes a relevant phone start ringing. When a user picks up the phone, signaling information to that effect is sent to the caller at the point A via the VoIP router  22 A and the PBX  21 A. The caller at the point A leans that his/her call is connected. 
   11. Audio communication is also converted into packets in the same manner as the signaling information, and these packets are exchanged between the VoIP routers. 
   12. When the user hangs up after finishing the call, the control unit  30  of the VoIP router  22 A on the caller side notifies the gatekeeper  24  of the end of the call. 
   13. The control unit  40  of the gatekeeper  24  resets the flag in the relevant table in response to the notice from the VoIP router  22 A. 
   This ends the communication between the point A and the point B. 
   A phone call from the point A to the point C is processed in much the same manner as described above, with the VoIP router  22 C taking a place of the VoIP router  22 B. 
   Concurrently with the audio communication, data communication can be conducted between the server  23  and the personal computer  25  of the point B or between the server  23  and the personal computer  25  of the point C. In practice, audio communication and data communication coexist as they are conducted. 
   The present invention improves efficiency of data communication while keeping constant the transmission intervals of audio packets for the purpose of securing audio quality. In order to keep constant the transmission intervals of audio packets, a long packet for data communication is evenly divided into packets of a predetermined length. The shorter the length of the data packets, the better the audio quality is. Improvement of audio quality comes at the expense of throughput of data communication. 
   In order to enhance efficiency of data communication while securing audio quality, therefore, the present invention adjusts a length that divides a long packet according to the procedure as follows. 
   [First Method] 
   This method determines a fragment size of data packets based on a wait time of an audio packet in queue where the wait time is measured by the VoIP router. 
   In  FIG. 4 , the routing unit  33  of the VoIP router  22 A creates a queue for each session. The routing unit  33  of the VoIP router  22 A measures a wait time of an audio packet in queue, and notifies the control unit  30  of the measured wait time. 
   The control unit  30  computes an average deviation from tens or hundreds of measurements, and adjusts a fragment size by following the procedure as shown in  FIG. 7 . 
     FIG. 7  is a flowchart of a method of adjusting a fragment size. 
   At a step S 10 , a check is made as to whether the deviation falls within a predetermined range. 
     FIG. 8  is an illustrative drawing for explaining how to determine the fragment size based on the deviation. 
   When the deviation continues to exceed a certain threshold (B) for more than a predetermined time period as shown in a time period T 2  in  FIG. 8 , the control unit  30  ascertains that the transmission intervals of audio packets fluctuates so much as to make it difficult to maintain audio quality. The control unit  30  instructs the routing unit  33  to make the fragment size smaller than a default size. The routing unit  33  reduces the MTU size, thereby making smaller the packet size by a factor of 0.X. This corresponds to a step S 11 . 
   When the deviation continues to stay within the predetermined range as shown in a time period T 3  in  FIG. 8 , the control unit  30  instructs the routing unit  33  to return the fragment size to the default size. The routing unit  33  returns the MTU size to the default size. This corresponds to a step S 12 . 
   When the deviation continues to fall below a certain threshold (A) for more than a predetermined time period as shown in a time period T 4  in  FIG. 8 , the control unit  30  ascertains that the transmission intervals of audio packets fluctuates so little as to warrant an increase of data throughput. The control unit  30  instructs the routing unit  33  to make the fragment size larger than the default size. The routing unit  33  enlarges the MTU size, thereby making larger the packet size by a factor of 1.X. This corresponds to a step S 13 . 
   As a result, data packets are divided by the default MTU size during the time periods T 1  and T 3  shown in  FIG. 8 , whereas data packets are divided by 0.X times the default MTU size during the time period T 2 , and are divided by 1.X times the default MTU size during the time period T 4 . 
   In this manner, the present invention can insure desired audio quality during the time period T 2 , and can improve data throughput during the time period T 4 . 
   In the above description, a deviation is obtained from measurements of a wait time of audio packets in queue, and, then, is compared with some thresholds. Alternatively, a wait time rather than the deviation may be used and compared with thresholds. 
   [Second Method] 
   This method determines a fragment size of data packets based on a delay time of a network where the delay time is measured by the VoIP router using a hello packet. 
   The control unit  30  of the VoIP router  22 A exchanges hello packets at constant intervals with other VoIP routers by using the routing protocol. 
   The control unit  30  measures a response time as a time period that passes from transmission of a hello packet to reception of the hello packet returning from another VoIP router, and adjusts a fragment size by following the procedure as shown in  FIG. 9 . 
     FIG. 9  is a flowchart of a method of adjusting a fragment size. 
   At a step S 10 , a check is made as to whether the response time falls within a predetermined range. 
   When the response time continues to exceed a certain threshold for more than a predetermined time period, the control unit  30  ascertains that a delay time of the network has increased to make it difficult to maintain audio quality. The control unit  30  instructs the routing unit  33  to make the fragment size smaller than a default size. The routing unit  33  reduces the MTU size, thereby making smaller the packet size. This corresponds to a step S 11 . 
   When the response time continues to stay within the predetermined range, the control unit  30  instructs the routing unit  33  to return the fragment size to the default size. The routing unit  33  returns the MTU size to the default size. This corresponds to a step S 12 . 
   When the response time continues to fall below a certain threshold for more than a predetermined time period, the control unit  30  ascertains that the delay time of the network has decreased to warrant an increase of data throughput. The control unit  30  instructs the routing unit  33  to make the fragment size larger than the default size. The routing unit  33  enlarges the MTU size, thereby making larger the packet size. This corresponds to a step S 13 . 
   As a result, data packets are divided by the default MTU size when the delay time of the network stays within the predetermined range. On the other hand, data packets are divided by smaller than the default MTU size when the delay time of the network is long, and are divided by larger than the default MTU size when the delay time of the network is short. 
   In this manner, the present invention can improve data throughput while insuring desired audio quality. 
   In the above description, the response time of the network is obtained from measurements of a time period that passes from transmission of audio packets to reception of the audio packets, and, then, is compared with some thresholds. Alternatively, a deviation of the response time may be obtained and compared with thresholds. 
   [Third Method] 
   This method determines a fragment size of data packets based on how many times a notice of network congestion is received. 
   In networks such as frame-relay networks, ATM networks, etc., when congestion occurs, the VoIP router  22 A is notified of the congestion. As the WAN-interface unit  32  of the VoIP router  22 A receives the notice of congestion, the WAN-interface unit  32  passes the notice to the control unit  30 . 
   In response, the control unit  30  of the VoIP router  22 A counts how many times the notice of congestion is received during a predetermined time period, and adjusts a fragment size by following the procedure as shown in  FIG. 10 . 
     FIG. 10  is a flowchart of a method of adjusting a fragment size. 
   At a step S 10 , a check is made as to whether the number of received congestion notices falls within a predetermined range. 
   When the number of received congestion notices continues to exceed a certain threshold for more than a predetermined time period, the control unit  30  ascertains that the network congestion has worsened to such an extent as to make it difficult to maintain audio quality. The control unit  30  instructs the routing unit  33  to make the fragment size smaller than a default size. The routing unit  33  reduces the MTU size, thereby making smaller the packet size. This corresponds to a step S 11 . 
   When the number of received congestion notices continues to stay within the predetermined range, the control unit  30  instructs the routing unit  33  to return the fragment size to the default size. The routing unit  33  returns the MTU size to the default size. This corresponds to a step S 12 . 
   When the number of received congestion notices continues to fall below a certain threshold for more than a predetermined time period, the control unit  30  ascertains that the network congestion is so little as to warrant an increase of data throughput. The control unit  30  instructs the routing unit  33  to make the fragment size larger than the default size. The routing unit  33  enlarges the MTU size, thereby making larger the packet size. This corresponds to a step S 13 . 
   As a result, data packets are divided by the default MTU size when the number of congestion notices stays within the predetermined range. On the other hand, data packets are divided by smaller than the default MTU size when the number of congestion notices is large, and are divided by larger than the default MTU size when the number of congestion notices is small. 
   In this manner, the present invention can improve data throughput while insuring desired audio quality. 
   In the above description, the number of received congestion notices is obtained by counting how many times the notice of congestion is received from the network, and, then, is compared with some thresholds. Alternatively, a deviation of the number of congestion notices may be obtained and compared with thresholds. 
   [Fourth Method] 
   This method determines a fragment size of data packets based on the number of audio calls that is reported from an apparatus that counts such a number. 
   The gatekeeper  24  can check the number of audio calls taking place at each VoIP router by referring to the communication flags provided in the gatekeeper table  41 . When the number of audio calls changes, the gatekeeper  24  notifies the control unit  30  of the number of audio calls. 
   In response, the control unit  30  of the VoIP router  22 A adjusts a fragment size based on the number of audio calls as shown in  FIG. 11 . 
     FIG. 11  is a flowchart of a method of adjusting a fragment size. 
   At a step S 10 , a check is made as to whether the number of calls falls within a predetermined range. 
   When the number of calls continues to exceed a certain threshold for more than a predetermined time period, the control unit  30  ascertains that the number of audio packets has increased to such a level as to make it difficult to maintain audio quality. The control unit  30  instructs the routing unit  33  to make the fragment size smaller than a default size. The routing unit  33  reduces the MTU size, thereby making smaller the packet size. This corresponds to a step S 11 . 
   When the number of calls continues stay within the predetermined range, the control unit  30  instructs the routing unit  33  to return the fragment size to the default size. The routing unit  33  returns the MTU size to the default size. This corresponds to a step S 12 . 
   When the number of calls continues to fall below a certain threshold for more than a predetermined time period, the control unit  30  ascertains that it is warranted to increase data throughput. The control unit  30  instructs the routing unit  33  to make the fragment size larger than the default size. The routing unit  33  enlarges the MTU size, thereby making larger the packet size. This corresponds to a step S 13 . 
   As a result, data packets are divided by the default MTU size when the number of calls stays within the predetermined range. On the other hand, data packets are divided by smaller than the default MTU size when the number of calls is large, and are divided by larger than the default MTU size when the number of calls is small. 
   As the number of audio calls that are simultaneously taking place increases, the number of audio packets increases. This makes it necessary to divide data packets into smaller fragments in order to maintain a desired audio quality. The fourth embodiment of the present invention changes the fragment size of data packets in response to the number of audio calls, thereby making it possible to improve data throughput while insuring desired audio quality. 
   [Fifth Method] 
   This method determines a fragment size of data packets based on the number of audio calls that is counted by the VoIP router. 
   The control unit  30  of the VoIP router  22 A can check the number of audio calls from the signaling information. The VoIP router  22 A lets the control unit  30  count the number of audio calls. The control unit  30  of the VoIP router  22 A adjusts a fragment size based on the number of audio calls as shown in  FIG. 12 . 
     FIG. 12  is a flowchart of a method of adjusting a fragment size. 
   At a step S 10 , a check is made as to whether the number of calls falls within a predetermined range. 
   When the number of calls continues to exceed a certain threshold for more than a predetermined time period, the control unit  30  ascertains that the number of audio packets has increased to such a level as to make it difficult to maintain audio quality. The control unit  30  instructs the routing unit  33  to make the fragment size smaller than a default size. The routing unit  33  reduces the MTU size, thereby making smaller the packet size. This corresponds to a step S 11 . 
   When the number of calls continues stay within the predetermined range, the control unit  30  instructs the routing unit  33  to return the fragment size to the default size. The routing unit  33  returns the MTU size to the default size. This corresponds to a step S 12 . 
   When the number of calls continues to fall below a certain threshold for more than a predetermined time period, the control unit  30  ascertains that it is warranted to increase data throughput. The control unit  30  instructs the routing unit  33  to make the fragment size larger than the default size. The routing unit  33  enlarges the MTU size, thereby making larger the packet size. This corresponds to a step S 13 . 
   As a result, data packets are divided by the default MTU size when the number of calls stays within the predetermined range. On the other hand, data packets are divided by smaller than the default MTU size when the number of calls is large, and are divided by larger than the default MTU size when the number of calls is small. 
   According to the fifth embodiment, the present invention changes the fragment size of data packets in response to the number of audio calls, thereby making it possible to improve data throughput while insuring desired audio quality. 
   [Detailed Operation] 
   In the following, details of operation of the VoIP router will be described with reference to the first method. 
     FIG. 6  is a block diagram of the VoIP router. 
   As previously described, the VoIP router includes the control unit  30 , the WAN-interface unit  32 , the routing unit  33 , the audio-interface unit  34 , and the LAN-interface unit  35 . 
   The routing unit  33  in  FIG. 6  includes a queue-wait-time-monitoring timer  50 , a packet- transmission unit  51 , a queue  52 , an IP-packet unit  53 , and a fragmentation unit  54 . 
   The queue-wait-time-monitoring timer  50  measures a wait time of an audio packet in queue, and sends the measurement to the control unit  30 . The packet-transmission unit  51  transmits audio packets ahead of other packets under the control of the control unit  30 . The queue  52  has data packets and audio packets waiting therein, and is provided for each session under the control of the control unit  30 . The IP-packet unit  53  converts audio signals into packets as the audio-interface unit  34  digitizes the audio signals. The fragmentation unit  54  divides data packets into fragments of a predetermined size under the control of the control unit  30 . 
   LAN-data packets are received by the LAN-interface unit  35  of the VoIP router  22 A, and are forwarded to the fragmentation unit  54  of the routing unit  33 . The fragmentation unit  54  breaks the packets into fragments of proper sizes, which are then sent to the queue  52 . There are a plurality of queues  52 , each of which is prioritized. In  FIG. 6 , for example, higher priority is given to the queues as the queues come closer to the bottom. In the order of priority, the packet-transmission unit  51  takes out packets from the queues  52 , and the WAN-interface unit  32  transmits these queues. 
   Packets each wait in the queues  52  until their turn comes. A time period during which a packet stays waiting in the queue is referred to as a wait time in queue. When audio is transmitted as packets, it is necessary to keep packet intervals constant in order to maintain audio quality. It is desirable, therefore, that a wait time in queue is as short and constant as possible. A need for a shorter wait time is satisfied by putting audio packets in the queue that is given priority. As for constancy, fluctuation of a waiting time in queue is determined by how often data having a packet length longer than audio packets are inserted between audio packets during transmission. 
   When the wait time in queue fluctuates violently, there is a need to shorten a fragment size of data packets. When the wait time in queue stays constant, on the other hand, it is reasonable to ascertain that audio quality is properly maintained, so that the fragment size is increased with an aim of improving data throughput. 
   Further, the present invention is not limited to these embodiments, but various variations and modifications may be made without departing from the scope of the present invention. 
   The present application is based on Japanese priority application No. 11-229468 filed on Aug. 13, 1999, with the Japanese Patent Office, the entire contents of which are hereby incorporated by reference.