JP2009272764A - Communication system, and call control apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a communication system and a call control apparatus, which can decide an optimal jitter buffer size for each communication terminal which performs speech communication and for each call. <P>SOLUTION: Each of at least two communication networks (200, 300) of the communication system 1 includes a communication terminal 2 with a memory 24 which temporarily stores coded data included in a plurality of received packets for collective decoding; and the call control apparatus 10. When an establishment request signal of a communication session to the communication terminal 2 from the communication terminal 3 is received, the call control apparatus 10 obtains information about a delay which occurs in another communication network 300 through which the call passes from another communication terminal 3 to the self communication network 200 to which the call control apparatus 10 belongs, obtains a data buffer size from the information and the information about a delay which occurs in the self communication network 200, and notifies it to the communication terminal 2. The communication terminal 2 sets a memory capacity of the memory 24 as the data buffer size, and starts communication afterwards. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present application relates to a communication system and a call control device, and particularly to a communication system and a call control device that perform real-time communication via an IP network.

  In recent years, various communication services using an IP network have been provided. As one of such communication services, there is an IP telephone service using a voice over internet protocol (VoIP). In the IP telephone service, one of two IP telephones communicating with each other divides a voice signal into a plurality of voice packets and transmits the voice packets to the other IP telephone. At that time, in principle, the IP network cannot always keep the interval at which these voice packets arrive at the receiving IP telephone. Therefore, the voice packet reception interval varies in the IP telephone on the receiving side. This variation in the reception interval is called jitter or delay jitter. If the delay jitter increases, the voice signal decoded by the IP telephone on the receiving side becomes intermittent, so that the call quality deteriorates.

In order to avoid deterioration in call quality caused by delay jitter, an IP telephone has a memory area called a jitter buffer that temporarily stores a predetermined amount of voice data. The IP telephone can obtain continuous audio signals by collectively decoding the audio data stored in the jitter buffer. However, if the memory size of the jitter buffer is too large, the period from when an audio signal is input through one IP telephone to when the audio signal is decoded by the other IP telephone becomes long. Therefore, a user who uses an IP telephone feels that there is a blank period every time the other party answers, like an old satellite telephone, and therefore cannot make a natural call.
In order to solve the above problem, a private branch exchange that sets an appropriate jitter buffer for each IP telephone has been developed (see, for example, Patent Document 1).

JP 2005-269134 A

  The private branch exchange disclosed in Patent Document 1 includes a first IP telephone existing on an IP network under the control of the private branch exchange and a second IP telephone connected via a wide area network (WAN). When a call is made between the two, the delay jitter during the call is measured. Then, the private branch exchange determines the size of the jitter buffer based on the measurement result of the delay jitter, and stores the size of the jitter buffer in association with the IP address of the second IP telephone. Thereafter, when a call is made again between the first IP phone and the second IP phone, the private branch exchange exchanges the second buffer by the size of the jitter buffer associated with the IP address of the second IP phone. Voice packets received from IP telephones are stored. When the amount of voice packets corresponding to the size of the jitter buffer is accumulated, the private branch exchange transmits the voice packets to the first IP telephone. For this reason, the private branch exchange can remove the delay jitter in the call between the first IP telephone and the second IP telephone.

  However, in the IP telephone service, in order to communicate a voice signal, an IP network in which a plurality of users share a band is used instead of a band-guaranteed network such as a circuit switching network. For this reason, the magnitude of delay jitter in the IP telephone service may vary depending on the usage status of the IP network during a call. Therefore, even if the size of the jitter buffer is determined based on the measurement result of the delay jitter during the past call, the size may not necessarily be optimal. In particular, when a plurality of IP networks are interposed between IP telephones that make a call, there is a possibility that the variation in delay jitter for each call will increase.

  Accordingly, an object of the present application is to provide a communication system and a call control device that can determine an optimum jitter buffer size for each communication terminal that performs a call and for each call.

  According to one embodiment, a communication system having at least two communication networks that transmit and receive encoded data in units of packets is provided. In such a communication system, each of at least two communication networks includes a memory that stores encoded data included in a plurality of received packets, and a signal processing unit that collectively decodes the encoded data stored in the memory. And a call control device that establishes a communication session between the first communication terminal and the second communication terminal connected to any one of at least two communication networks. Then, when the call control device receives a communication session establishment request signal for the first communication terminal from the second communication terminal, the call control device passes between the second communication terminal and the own communication network to which the call control device belongs. Other network delay information related to a delay generated in another communication network of at least two communication networks is acquired from the other communication network, and the data buffer is obtained from the other network delay information and the own network delay information related to a delay generated in the own communication network. The size is obtained and notified to the first communication terminal. The first communication terminal sets the memory capacity of the memory to the data buffer size notified from the call control device, and then starts communication.

  According to another embodiment, the first communication terminal is connected to the first communication network among the plurality of communication networks that transmit and receive encoded data in units of packets, and is connected to the first communication network. And a call control device for establishing a communication session between second communication terminals connected to any one of the plurality of communication networks. When the call control apparatus receives a request signal for establishing a communication session between the second communication terminal and the first communication terminal, the call control apparatus transmits another signal from the second communication terminal to the first communication network. The other network delay information related to the delay occurring in the communication network is acquired from the other communication network, and the first communication terminal uses the other network delay information and the own network delay information regarding the delay generated in the first communication network to A data buffer size for storing encoded data included in a packet received from the terminal is obtained and notified to the first communication terminal.

  The communication system and the call control device disclosed in the present application have an effect that an optimum jitter buffer size can be determined for each communication terminal that performs a call and for each call.

Hereinafter, a communication system according to an embodiment will be described with reference to the drawings.
In a communication system according to one embodiment, a plurality of IP networks are connected via a router. An IP telephone is connected to each IP network. Each IP network has a call control server for establishing a communication session when the IP telephone performs voice communication with another IP telephone. When a call control server establishes a communication session between IP telephones along communication paths that pass through a plurality of IP networks, information on communication delay of the IP network under its management is placed on the communication path. Notify the call control server of some other IP network. The call control server also receives information regarding the delay from other call control servers on the communication path. Then, the call control server determines an optimum jitter buffer size for each communication session based on information on delays of all IP networks existing on the communication path. Then, the call control server notifies the IP telephone that performs voice communication in the communication session of the optimum buffer size. On the other hand, the IP telephone sets a storage capacity of the jitter buffer based on the buffer size notified from the call control server, and then starts a call.

  FIG. 1 is a schematic configuration diagram of a communication system 1 according to one embodiment. As shown in FIG. 1, the communication system 1 includes a telephone exchange network 100, a first IP network 200, and a second IP network 300. The first IP network 200 and the second IP network 300 include the call control server 10 and the call control server 20, respectively. A plurality of subscriber phones are connected to the telephone exchange network 100. On the other hand, a plurality of IP telephones are connected to the first IP network 200 and the second IP network 300, respectively. For the sake of simplicity, FIG. 1 shows only the telephone 4 as a subscriber telephone connected to the telephone exchange network. Similarly, in FIG. 1, only the IP telephone 2 is shown as the IP telephone connected to the first IP network 200, and only the IP telephone 3 is shown as the IP telephone connected to the second IP network 300. The telephone exchange network 100 and the first IP network 200 are connected via a gateway 50. The first IP network 200 and the second IP network 300 are connected via the router 51. In FIG. 1, only two IP networks are shown for simplicity. However, the communication system 1 may include three or more independent IP networks, and each IP network may be connected to each other via a router or a gateway. Also in this case, at least one call control server is installed for each IP network.

The telephone exchange network 100 is a network that establishes a dedicated communication line between subscriber telephones connected to the telephone exchange network 100 by an exchange and enables a call between the subscriber telephones. The telephone exchange network 100 can be, for example, an existing public exchange telephone network connecting a fixed telephone, an existing mobile communication network connecting a mobile telephone such as a mobile phone, or a communication network combining them.
The telephone 4 is a telephone used in a public switched telephone network or a mobile communication network such as a fixed telephone, a mobile phone, and a PHS.

  The gateway 50 is a format suitable for the network to which each telephone is connected, and signals transmitted between the telephone 4 connected to the telephone switching network 100 and the IP telephone 2 or IP telephone 3 connected to the IP network. Convert to a signal. When communication is performed between the telephone 4 and the IP telephone 2 or 3, the gateway 50 functions as a subscriber telephone instead of the IP telephone 2 or 3 with respect to the telephone exchange network 100. On the other hand, the gateway 50 functions as an IP telephone instead of the telephone 4 with respect to the IP network 200. The gateway 50 enables a telephone call between the telephone 4 and the IP telephone 2 or the IP telephone 3. Note that, for example, various known gateway devices can be used as the gateway 50, and therefore, detailed description of the configuration and functions of the gateway 50 are omitted.

The first IP network 200 and the second IP network 300 are communication networks that perform communication between communication terminals by exchanging data in units of IP packets according to the Internet protocol (IP). For example, the first IP network 200 includes a plurality of communication terminals that transmit and receive data in units of IP packets in accordance with the Internet protocol, a plurality of transmission lines such as conducting wires or optical fiber cables for transmitting IP packets, and the transmission lines. And a plurality of routers and switching hubs for transferring IP packets to a predetermined destination according to the header information. The second IP network 300 also has the same configuration as the first IP network 200.
In addition, the first IP network 200 and the second IP network 300 cause delay and delay jitter when transmitting a packet, respectively. The values of the delay and delay jitter are estimated or continuously measured according to the configuration of the IP network and the assumed traffic. Furthermore, the allowable delay times of the IP networks 200 and 300 are set according to the communication quality guaranteed by the IP networks 200 and 300, respectively. The delay, delay jitter, and allowable delay time that occur in the IP networks 200 and 300 are stored in the call control servers 10 and 20 that manage the IP networks 200 and 300, respectively.

The first IP telephone 2 and the second IP telephone 3 that are communication terminals are telephones that perform voice communication using voice communication technology using an Internet protocol such as VoIP. Since the first IP telephone 2 and the second IP telephone 3 have the same functions and configurations, the first IP telephone 2 will be described below.
FIG. 2 shows a schematic configuration diagram of the IP telephone 2. The IP telephone 2 includes a voice processing unit 21, a packet transmission / reception unit 22, and a call control unit 23. Each of these units is a functional module implemented by a control circuit such as a CPU included in the IP telephone 2 and a program executed on the control circuit. Or each said part may be mounted in the IP telephone 2 as firmware. The IP telephone 2 has a jitter buffer 24 having a semiconductor memory such as a RAM. The jitter buffer 24 is a memory that temporarily stores an encoded audio signal included in the received IP packet, and its storage capacity can be freely set below the physical maximum storage capacity. Furthermore, the IP telephone 2 has a microphone 25 and a speaker 26.

The voice processing unit 21 compresses the voice signal using any one of various compression coding techniques in order to transmit the voice signal collected from the IP telephone 2 through the microphone 25 to another telephone. To do. For example, the audio processing unit 21 can select and use a compression coding technique defined as G711, G729a, G723.1, or the like. Then, the voice processing unit 21 passes the compression-coded voice signal to the packet transmitting / receiving unit 22.
The voice processing unit 21 decodes the compression-coded voice signal received from the other telephone set via the packet transmission / reception unit 22. The audio processing unit 21 performs echo canceller processing on the audio signal, and then converts the audio signal into audio PCM data in the PCM format. Then, the audio processing unit 21 passes the audio PCM data to the speaker 26 and reproduces the audio signal.

The packet transmitting / receiving unit 22 divides the compression-coded audio signal received from the audio processing unit 21 into data in units of IP packets. Then, the packet transmitting / receiving unit 22 creates a plurality of IP packets by adding header information including the IP address of the other party of the call to the divided data. Then, the packet transmitting / receiving unit 22 sends these IP packets to the IP network 200 via a communication interface that operates according to a communication protocol such as TCP / IP or UDP / IP.
On the other hand, when receiving an IP packet including a voice signal transmitted from another telephone (including an IP telephone and a subscriber telephone), the packet transmitting / receiving unit 22 extracts the voice signal from the IP packet and stores it in the jitter buffer 24. Remember. Then, when the audio signal accumulated in the jitter buffer 24 becomes full of the storage capacity set in the jitter buffer 24, the packet transmitting / receiving unit 22 passes all the audio signals accumulated in the jitter buffer 24 to the audio processing unit 21. Alternatively, the packet transmitting / receiving unit 22 may store the IP packet itself including the audio signal in the jitter buffer 24. In this case, when the storage capacity set in the jitter buffer 24 becomes full, the packet transmitting / receiving unit 22 extracts a voice signal from all IP packets stored in the jitter buffer 24 and passes it to the voice processing unit 21.

  The call control unit 23 establishes a communication session with the other party of the call or terminates the communication session according to the call control protocol. In this embodiment, a session establishment protocol (SIP) is used as the call control protocol. When the communication control unit 23 establishes a communication session from the IP telephone set 2 to another telephone set, the call control unit 23 executes a calling process. For example, the call control unit 23 designates the destination telephone address and sends an Invite signal to the call control server 10 that manages the IP network 200 to which the IP telephone 2 is connected. On the other hand, when the call control unit 23 is requested to establish a communication session from another telephone set and receives an Invite signal from the call control server 10, the call control unit 23 performs an incoming call process, and performs a 100try signal, a 180 Ringing signal, a 200 OK signal, a 300 Multiple Choices signal, 400 Bad A response signal such as a Request signal is returned to the call control server 10.

The call control unit 23 sets the storage capacity of the jitter buffer 24 every time a communication session is established. FIG. 3 shows an operation flowchart of the call controller 23 for the storage capacity setting process of the jitter buffer.
The call control unit 23 starts the jitter buffer storage capacity setting process after the session establishment process is started. The call control unit 23 includes the jitter size information indicating the optimum jitter buffer size in the control signal (for example, Invite signal, 180 Ringing signal, 200 OK signal) received from the call control server 10 for session establishment. It is determined whether or not (step S301). The jitter size information will be described later. If the control signal does not include jitter size information, the call control unit 23 sets the storage capacity of the jitter buffer 24 to a predetermined initial value (step S302). For example, the initial value can be a storage capacity necessary for storing a voice signal in a period (for example, 50 msec) corresponding to delay jitter assumed when voice communication is performed in the IP network 200.

On the other hand, if jitter size information is included in the control signal in step S301, the call control unit 23 compares the jitter buffer size indicated in the jitter size information with the initial value of the jitter buffer size (step S303). ). If the jitter buffer size indicated in the jitter size information is different from the initial value of the jitter buffer size, the call control unit 23 sets the storage capacity of the jitter buffer 24 to the jitter buffer size indicated in the jitter size information. The capacity corresponding to the audio signal is set (step S304). On the other hand, when the jitter buffer size indicated in the jitter size information is equal to the initial value of the jitter buffer size in step S304, the call control unit 23 sets the storage capacity of the jitter buffer 24 to the initial value (step S304). S302). After step S302 or S304, the call control unit 23 ends the storage capacity setting process of the jitter buffer 24.
As described above, since the IP telephone 2 sets the storage capacity of the jitter buffer 24 according to the jitter buffer size notified from the call control server when the communication session is established, the storage capacity of the jitter buffer 24 is set for each communication session. An optimal value can be obtained.

  The first call control server 10 and the second call control server 20 which are call control devices control a communication session between IP telephones or between an IP telephone and the telephone 4 according to a call control protocol. When one communication terminal is the telephone 4, the first call control server 10 and the second call control server 20 use the gateway 50 instead of the telephone 4 as a communication terminal to perform processing such as establishment of a communication session. I do. Then, the gateway 50 performs processing such as establishment of a communication session in the telephone exchange network 100 instead of the IP telephone 2 or 3. In the present embodiment, as described above, the SIP protocol is used as the call control protocol. In addition, since the 1st call control server 10 and the 2nd call control server 20 have the same structure and function, below, the 1st call control server 10 is demonstrated.

  FIG. 4 is a schematic configuration diagram of the call control server 10. The call control server 10 includes a connection processing unit 31 and a recommended buffer size determination unit 32. Each of these units is a functional module implemented by a program executed on a control circuit such as a CPU included in the call control server 10. Alternatively, each of the above units may be implemented in the call control server 10 as firmware. The call control server 10 includes a communication unit 33 having a communication interface for the IP network and its peripheral circuits. Further, the call control server 10 includes a storage unit 34 having a semiconductor memory such as a RAM or a magnetic recording medium and an access device thereof. The storage unit 34 includes a routing table indicating the location information of the IP telephone connected to the IP network, an estimated value or a measured value of delay and delay jitter generated in the IP network 200 managed by the call control server 10, and an allowable value. Memorize delay time.

  The connection processing unit 31 has a function as a proxy server. Therefore, the connection processing unit 31 executes processing related to establishment, change, or termination of a communication session in response to a request from the IP telephone 2 or another call control server (for example, the call control server 20). For example, when the connection processing unit 31 receives a communication session establishment request (Invite signal) from the IP telephone 2 to another IP telephone or a subscriber telephone via the communication unit 33, the routing table stored in the storage unit 34. To determine the signal transfer destination. Then, the connection processing unit 31 transmits an Invite signal to the IP telephone of the transfer destination or another call control server via the communication unit 33 and returns a 100 try signal to the IP telephone 2. In addition, when the connection processing unit 31 receives a response signal such as a 180 Ringing signal or a 200 OK signal from the transfer destination IP telephone or another call control server via the communication unit 33, the connection processing unit 31 communicates a response signal corresponding to the received response signal. The data is transmitted to the IP telephone 2 via the unit 33.

  Further, the connection processing unit 31 adds information on the delay and delay jitter obtained by the recommended buffer size determining unit 32 to a response signal such as an Invite signal or 180 Ringing signal transmitted to the IP telephone or another call control server.

  The recommended buffer size determination unit 32 is based on information on delay and delay jitter generated in the IP network 200 managed by the call control server 10 and delay and delay jitter included in an Invite signal received from another call control server. Thus, the optimum jitter buffer size is calculated for one or more IP networks through which the communication session is to be established. Then, the recommended buffer size determination unit 32 passes these values to the connection processing unit 31. Details of the calculation of the optimum jitter buffer size will be described later.

  FIG. 5 shows a sequence when a call is requested from IP phone 2 to IP phone 3 as an example of a sequence for establishing a communication session. As shown in FIG. 5, first, the IP telephone 2 transmits an Invite signal to the call control server 10 that manages the IP network 200 to which the IP telephone 2 is connected (step S501). The call control server 10 refers to the routing table stored in the storage unit 34 by the connection processing unit 31 and manages the IP network 300 to which the IP telephone 3 is connected as the next transmission destination of the Invite signal. The server 20 is determined. The call control server 10 edits the Invite signal so that the Invite signal includes the delay and delay jitter of the IP network 200 managed by the call control server 10 (step S502). Then, the call control server 10 transmits an Invite signal to the call control server 20 (step S503). Further, the call control server 10 returns a 100 try signal to the IP telephone 2 in order to notify the IP telephone 2 that the establishment of the communication session is being attempted.

  The call control server 20 refers to the delay and delay jitter included in the Invite signal received from the call control server 10 by the recommended buffer size determination unit 32 and the delay and delay jitter in the IP network 300 to optimize the IP telephone 3. Determine the correct jitter buffer size. Then, the call control server 20 uses the connection processing unit 31 to add jitter size information indicating the optimum jitter buffer size to the Invite signal (step S504). Then, the call control server 20 transmits an Invite signal to the IP telephone 3 (Step S505). The call control server 20 returns a 100 try signal to the call control server 10.

  When the IP telephone 3 receives the Invite signal from the call control server 20, the call control unit 23 refers to the jitter size information included in the Invite signal and sets the storage capacity of the jitter buffer 24 (step S506). Thereafter, the IP telephone 3 returns a 100 try signal to the call control server 20. Furthermore, when the predetermined incoming call processing is completed, the IP telephone 3 transmits a 180 Ringing signal indicating that the call is being made to the call control server 20 (step S507).

  When the call control server 20 receives the 180 Ringing signal from the IP telephone 3, the connection processing unit 31 adds the jitter size information obtained in Step S504 to the 180 Ringing signal (Step S508). The call control server 20 also adds delay information of each IP network through this communication session to the 180 Ringing signal. Then, the call control server 20 transmits a 180 Ringing signal to the call control server 10 (step S509). When the call control server 10 receives the 180 Ringing signal from the call control server 20, the recommended buffer size determination unit 32 refers to the jitter size information included in the 180 Ringing signal to determine the optimum jitter buffer size for the IP telephone 2. To do. Then, the call control server 10 uses the connection processing unit 31 to add jitter size information indicating the optimum jitter buffer size to the 180 Ringing signal (step S510). Then, the call control server 10 transmits the 180Ringing signal to the IP telephone 2 (step S511). When the IP telephone 2 receives the 180 Ringing signal from the call control server 10, the call control unit 23 refers to the jitter size information included in the 180 Ringing signal and sets the storage capacity of the jitter buffer 24 (step S512).

  Thereafter, when the IP telephone set 3 succeeds in the communication session establishment request from the IP telephone set 2, it transmits a 200 OK signal to the call control server 20. The call control server 20 transmits a 200 OK signal to the call control server 10, and the call control server 10 transmits a 200 OK signal to the IP telephone 2. When the IP telephone 2 receives the 200OK signal, it transmits an Ack signal to the IP telephone 3 in order to confirm that the communication session has been successfully established. Thereafter, transmission / reception of voice packets between the IP telephone 2 and the IP telephone 3 is started.

The operation of the connection processing unit 31 and the recommended buffer size determination unit 32 when editing the Invite signal (that is, the processing in steps S502 and S504 above) will be described with reference to the operation flowchart shown in FIG.
First, when the call control server receives the Invite signal, it is determined whether or not the IP telephone of the caller indicated by the Invite signal is connected to the IP network managed by the call control server (step S601). ). When the source IP telephone is not connected to the managed IP network, the recommended buffer size determining unit 32 manages the call destination, that is, the IP telephone requested to start the communication session, by the call control server. It is determined whether or not it is connected to the current IP network (step S602). When the called IP telephone is connected to the managed network, the recommended buffer size determination unit 32 estimates the maximum delay time that occurs in a plurality of IP networks that are routed in the communication session to be established. Then, the recommended buffer size determining unit 32 determines whether or not the maximum delay time is within the allowable delay time of the managed IP network (step S603). The recommended buffer size determination unit 32 reads the delay and delay jitter of the managed IP network from the storage unit 34 in order to obtain the maximum delay time. Also, the recommended buffer size determination unit 32 reads out the delay and delay jitter of each of the other IP networks from the received Invite signal through which the Invite signal has reached the call control server from the originating IP telephone. Then, the recommended buffer size determination unit 32 calculates the network delay time by adding the delay and delay jitter of the managed IP network and the delay and delay jitter of the other IP networks. Further, the recommended buffer size determination unit 32 adds the codec time required for compressing and encoding the audio signal and the time required for packetizing the encoded audio signal to the network delay time, and sets the maximum delay time. Ask.

  When the maximum delay time is within the allowable delay time of the managed IP network, the recommended buffer size determination unit 32 determines that each of the other IPs through which the Invite signal has passed from the caller IP phone to the call control server. The total delay jitter is calculated by adding the delay jitter of the network and the delay jitter of the managed IP network. Then, the recommended buffer size determining unit 32 sets the optimum jitter buffer size of the called IP telephone in the communication session to the total delay jitter. The recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. Then, the connection processing unit 31 writes the optimum jitter buffer size as jitter size information in the Invite signal transmitted to the called IP telephone (step S604).

  On the other hand, in step S603, when the maximum delay time exceeds the allowable delay time of the managed IP network, the recommended buffer size determining unit 32 sets the optimum jitter buffer size to the allowable jitter jitter in the managed IP network. Set to the maximum value. The maximum value of delay jitter is set in advance based on the call quality compensated by the managed IP network and stored in the storage unit 34. The recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. Then, the connection processing unit 31 writes the optimum jitter buffer size as jitter size information in the Invite signal transmitted to the called IP telephone (step S605).

  In step S602, if the called IP telephone is not connected to the managed IP network, that is, if it is connected to another IP network, the recommended buffer size determination unit 32 manages the management from the storage unit 34. Read the network delay and delay jitter below. Then, the recommended buffer size determination unit 32 notifies the connection processing unit 31 of the delay and the delay jitter. The connection processing unit 31 edits the Invite signal to the next transfer destination call control server, and adds the delay and delay jitter of the managed IP network (step S606).

  In step S601, if the source IP telephone is connected to the managed IP network, the recommended buffer size determining unit 32 determines whether the called IP telephone is connected to the same managed IP network. It is determined whether or not (step S607). When the called IP telephone is connected to the managed IP network, the communication session passes only through the IP network managed by the call control server. Therefore, the recommended buffer size determination unit 32 reads the delay jitter of the managed IP network from the storage unit 34, and sets the delay jitter as the optimum jitter buffer size. Then, the recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. The connection processing unit 31 writes the jitter buffer size in the jitter size information of the Invite signal transmitted to the called IP telephone (step S608).

On the other hand, if the called IP telephone is not connected to the managed IP network in step S607, the recommended buffer size determining unit 32 reads the delay and delay jitter of the managed IP network from the storage unit. Then, the recommended buffer size determination unit 32 notifies the connection processing unit 31 of the delay and the delay jitter. The connection processing unit 31 edits the Invite signal to the next transfer destination call control server, and adds the delay and delay jitter of the managed IP network (step S609).
After steps S604 to S606, S608, or S609, the connection processing unit 31 finishes editing the Invite signal, and transmits the Invite signal to the IP telephone or another call control server.

The operations of the connection processing unit 31 and the recommended buffer size determination unit 32 when editing the 180 Ringing signal to be transmitted to the originating IP telephone with reference to the operation flowchart shown in FIG. 7 (that is, the processing in step S510 above) Will be explained. For the 180 Ringing signal to be transmitted to other call control servers, the connection processing unit 31 adds the jitter size information included in the received 180 Ringing signal and the delay information of each IP network via this communication session to the 180 Ringing signal. To do.
First, when the call control server receives the 180Ringing signal, it is determined from the 180Ringing signal whether the called IP telephone is connected to the IP network managed by the call control server (step S701). When the called IP phone is connected to the managed IP network, the originating IP phone is also connected to the managed IP network, so the communication session is only the IP network managed by the call control server. Via. Therefore, the recommended buffer size determination unit 32 reads the delay jitter of the managed IP network from the storage unit 34, and sets the delay jitter as the optimum jitter buffer size. Then, the recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. The connection processing unit 31 writes the jitter buffer size in the jitter size information of the 180 Ringing signal transmitted to the called IP telephone (step S702).

  On the other hand, in step S701, when the called IP telephone is connected to another IP network instead of the managed IP network, the recommended buffer size determination unit 32 includes the jitter size information in the received 180 Ringing signal. It is determined whether or not (step S703). If the received 180Ringing signal does not include jitter size information, the connection processing unit 31 creates a 180Ringing signal to be transmitted to the originating IP telephone without including the jitter size information (step S704).

  In step S703, if the received 180Ringing signal includes jitter size information, the recommended buffer size determination unit 32 sets the buffer size of the delay jitter indicated in the jitter size information to each of the communication sizes that pass through this communication session. The network delay time is calculated by adding the delay of the IP network. Further, the recommended buffer size determination unit 32 adds the codec time required for compressing and encoding the audio signal and the time required for packetizing the encoded audio signal to the network delay time, and sets the maximum delay time. Ask. Then, the recommended buffer size determination unit 32 determines whether or not the maximum delay time is within the allowable delay time of the managed IP network (step S705).

  When the maximum delay time is within the allowable delay time of the managed IP network, the recommended buffer size determination unit 32 includes the optimum jitter buffer size of the source IP telephone in the communication session in the received 180 Ringing signal. Set to the value indicated in the jitter size information. The recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. Then, the connection processing unit 31 writes the optimum jitter buffer size as jitter size information in the 180 Ringing signal transmitted to the source IP telephone (step S706).

On the other hand, when the maximum delay time exceeds the allowable delay time of the managed IP network in step S705, the recommended buffer size determining unit 32 sets the optimum jitter buffer size to the allowable delay jitter in the managed IP network. Set to the maximum value. The recommended buffer size determination unit 32 notifies the connection processing unit 31 of the optimum jitter buffer size. Then, the connection processing unit 31 writes the optimum jitter buffer size as jitter size information in the 180 Ringing signal transmitted to the source IP telephone (step S707).
After step S702, S706, or S707, the connection processing unit 31 finishes editing the 180Ringing signal and transmits the 180Ringing signal to the originating IP telephone.

  In the flowcharts shown in FIGS. 6 and 7, the recommended buffer size determination unit 32 determines whether or not the source and called IP telephones are connected to the IP network managed by the call control server. Judge as follows. That is, the recommended buffer size determination unit 32 obtains the addresses of the caller and callee IP phones included in the Invite signal or 180Ringing signal. Then, the recommended buffer size determining unit 32 refers to the routing table storing the location information of the IP phone based on the obtained IP phone address, so that the IP phone is connected to the managed IP network. It can be determined whether it is connected to another IP network.

  FIG. 8 shows an example of an Invite signal (Invite signal edited in step S606 or S609 described above) transmitted to a call control server that manages another IP network. In this embodiment, since SIP is used as the call control protocol, all control signals such as the Invite signal and the 180 Ringing signal are text-based signals.

  As shown in FIG. 8, the Invite signal 800 is based on SIP, such as information indicating that the signal is an Invite signal, information indicating the address of the IP phone of the caller, and information indicating the address of the IP phone of the callee. Standard information 810 is included as an Invite signal. Furthermore, the Invite signal 800 includes delay information 811 and delay jitter information 812 generated in the IP network 200 managed by the call control server 10. In this embodiment, the delay information 811 includes a header notation “Network-Delay:” and a delay (msec unit) described subsequently. Similarly, the delay jitter information 812 includes a header notation “Network-Jitter:” followed by delay jitter (msec units) described subsequently. When a communication session to be established passes through a plurality of IP networks, the delay information 811 and the delay jitter information 812 describe the delay and delay jitter information of each IP network. For example, when the call control server 10 receives an Invite signal from another call control server, the delay information 811 and the delay jitter information 812 included in the Invite signal are “Network-Delay: 80” and “Network-Jitter”, respectively. : 50 ”. Assume that the delay occurring in the IP network 200 managed by the call control server 10 is 50 msec and the delay jitter is 30 msec. In this case, the connection processing unit 31 corrects the delay information 811 and the delay jitter information 812 as “Network-Delay: 80; 50” and “Network-Jitter: 50; 30”, respectively. Note that the connection processing unit 31 may describe the total delay value and the total delay jitter generated in each IP network as the delay information 811 and the delay jitter information 812. For example, in the case of the above example, the delay information 811 and the delay jitter information 812 are described as “Network-Delay: 130” (80 msec + 50 msec = 130 msec) and “Network-Jitter: 80” (50 msec + 30 msec = 80 msec), respectively. Is done.

  FIG. 9 shows an example of an Invite signal (Invite signal edited in step S604, 605 or S608 described above) transmitted to the IP telephone connected to the IP network managed by the call control server. As shown in FIG. 9, the Invite signal 900 is based on SIP, such as information indicating that the signal is an Invite signal, information indicating the address of the IP phone of the caller, and information indicating the address of the IP phone of the callee. Standard information 910 is included as an Invite signal. Further, the Invite signal 900 includes jitter size information 911 indicating an optimum jitter buffer size to be set in the IP telephone (that is, the called IP telephone) requested to establish a communication session. In the present embodiment, the jitter size information 911 includes a header notation “Jitter-Recommended-value:” followed by a jitter buffer size (msec unit) described subsequently.

  FIG. 10 shows an example of the 180 Ringing signal transmitted to the IP telephone in this embodiment. As shown in FIG. 10, the 180Ringing signal 1000 is based on SIP, such as information indicating that the signal is a 180Ringing signal, information indicating the address of the originating IP phone, and information indicating the address of the called IP phone. Standard information 1010 is included as a 180 Ringing signal. Further, the 180 Ringing signal 1000 includes jitter size information 1011 indicating an optimum jitter buffer size to be set in the IP telephone that has requested establishment of a communication session (that is, the IP telephone of the transmission source). In the present embodiment, as with the jitter size information 911, the jitter size information 1011 includes a header notation “Jitter-Recommended-value:” followed by a jitter buffer size (unit: msec). In addition, the 180 Ringing signal transmitted to the call control server further includes delay information (for example, “Network-Delay: 80; 50”) described with respect to the Invite signal.

  The call control server 10 may further include a registration processing unit that determines the address (that is, SIP URI) of the IP telephone requested to be registered and registers the current position of the IP telephone in the routing table. Good. Further, the call control server 10 has a function as a SIP redirect server (that is, when a request is received, the call control server 10 has a redirect unit that notifies the destination IP telephone of the destination of the IP telephone of the request). It may be. These registration processing unit and redirect unit can be a functional module or firmware implemented by a control circuit such as a CPU included in the call control server 10 and a program executed on the control circuit.

As described above, in the communication system according to one embodiment, when each call control server that manages different IP networks establishes a communication session between IP telephones, The delay and delay jitter are notified to other call control servers. Therefore, the call control server that manages the IP network to which the source or called IP telephone is connected, considers the delay and delay jitter of other IP networks every time a communication session is established, and the optimum jitter buffer. The size can be determined. In addition, since an IP telephone set included in the communication system can obtain an optimum jitter buffer size from a call control server that manages an IP network to which the IP telephone is connected, an optimum jitter buffer size for each communication session. Can be set.
Furthermore, the call control server uses the control signal used for establishing the communication session when the communication session is established, that is, before the call is started, using the delay information, the delay jitter information, or the jitter size information. Or notify other call control servers. Since the time required for the IP telephone to set the jitter buffer size is almost negligible, the communication system sets the optimum jitter buffer size without causing discomfort to the user who uses the IP telephone. be able to.

The communication system according to one embodiment has been described above, but the present invention is not limited to the above embodiment. For example, the call control unit 23 of the called IP telephone (that is, the IP telephone 3 in the sequence shown in FIG. 5) sets the storage capacity setting of the jitter buffer 24 until the Ack signal is received after the Invite signal is received. It may be executed at any timing. Further, the call control server transmits the delay information, delay jitter information, and jitter size information shown in FIGS. 8 to 10 to another call control server or IP by using a control signal different from the control signal defined by the call control protocol. You may notify a telephone. Furthermore, each call control server adds the IP network quality class information (class A, B or C) managed by itself instead of adding the delay of the IP network managed by itself to the above delay information. Also good. In this case, the recommended buffer size determination unit 32 of the call control server uses the maximum value of end-to-end delay guaranteed in the quality class information (for example, 100 msec in the case of class A) as the delay of the other IP network. Used to determine the correct jitter buffer size. Further, the connection processing unit 31 of the call control server may add the jitter size information to another response signal such as a 200OK signal instead of adding the 180 Ringing signal.
Furthermore, in step S706 of the 180 Ringing signal transmitted to the caller IP telephone shown in FIG. 7 above, the call control server again adds the delay jitter of all IP networks that pass through the communication session. Total delay jitter may be calculated. Then, the call control server may add the total delay jitter as the optimum jitter buffer size to the jitter size information of the 180 Ringing signal. In this case, in order to calculate the total delay jitter, the delay jitter information of each IP network is included in the 180 Ringing signal transmitted from the call control server on the call destination IP phone side to the call control server on the caller IP phone side. Added.

  Further, the call control server may notify the recommended compression coding technique instead of notifying the IP telephone of the optimal jitter buffer size or together with the notification of the optimal jitter buffer size. In this case, the call control server first performs compression coding technology with the best sound quality (that is, the longest time required for compression coding) in steps S603 and S705 for determining the optimum jitter buffer size. The maximum delay time is calculated using the codec time corresponding to. Then, the call control server determines whether or not the maximum delay time exceeds the allowable delay time of the managed IP network. When the maximum delay time is within the allowable delay time, the call control server notifies the IP telephone of the compression encoding technique with the best sound quality as the recommended compression encoding technique. On the other hand, if the maximum delay time exceeds the allowable delay time, the call control server calculates the maximum delay time again using the codec time corresponding to the next compression coding technology with the best sound quality and Judge whether or not. If the maximum delay time is within the allowable delay time, the call control server notifies the IP telephone as the recommended compression coding technique of the next compression coding technique with the best sound quality. On the other hand, when the maximum delay time exceeds the allowable delay time, the call control server calculates the maximum delay time again using the codec time corresponding to the third compression coding technology with the best sound quality, and exceeds the allowable delay time. It is determined whether or not. Thereafter, the call control server repeats the same processing until the maximum delay time is within the allowable delay time of the managed IP network or until the above processing is completed for all compression coding technologies that can be used by the IP telephone. . If the maximum delay time obtained using the compression coding technique with the shortest codec time out of the available compression coding techniques exceeds the allowable delay time, the call control server will compress the codec with the shortest codec time. The IP telephone is notified of the encoding technique as a recommended compression encoding technique. Furthermore, when notifying the IP telephone also of the optimum jitter buffer size, the call control server sets the optimum jitter buffer size to the maximum value of delay jitter allowed in the managed IP network. When the IP telephone is notified of the recommended compression encoding technique from the call control server, the IP telephone can select and use the compression encoding technique to be used based on the notification.

As described above, the call control server can evaluate the delay and delay jitter of each IP network through the communication session for each communication session and recommend an appropriate compression coding technique. Further, when the IP telephone uses the recommended compression coding technique, the communication system can use the optimum compression coding technique for each communication session. Therefore, the user can make a call with good sound quality when there is a sufficient amount of communication in the IP network. In addition, even when the IP network has no traffic, the user can make a call without intermittent voices or feeling a blank period.
Further, such a communication system can be used not only for voice calls but also for other types of communication that require real-time performance, for example, services for transmitting live video.
As described above, those skilled in the art can make various modifications in accordance with the embodiment to be implemented within the scope of the present invention.

The following supplementary notes are further disclosed regarding the embodiment described above and its modifications.
(Appendix 1)
A communication system having at least two communication networks for transmitting and receiving encoded data in units of packets,
Each of the at least two communication networks is
A first communication terminal having a memory for storing encoded data included in a plurality of received packets, and a signal processing unit for collectively decoding the encoded data stored in the memory;
A call control device for establishing a communication session between the first communication terminal and a second communication terminal connected to one of the at least two communication networks;
When the call control apparatus receives a communication session establishment request signal for the first communication terminal from the second communication terminal, the call control apparatus communicates from the second communication terminal to the own communication network to which the call control apparatus belongs. Other network delay information related to a delay that occurs in another communication network among the at least two communication networks that pass between them is acquired from the other communication network, and the other network delay information and a delay that occurs in the own communication network Determining the data buffer size from its own network delay information and notifying the first communication terminal;
The first communication terminal sets the storage capacity of the memory to the data buffer size notified from the call control device, and then starts communication.
Communications system. (1) (Figure 1)
(Appendix 2)
The other network delay information includes delay jitter that occurs in each of the other communication networks, and the own network delay information includes delay jitter that occurs in the own communication network,
The call control device calculates a total delay jitter by summing a delay jitter generated in each of the other communication networks and a delay jitter generated in the own communication network, and uses the total delay jitter as the data buffer size. The communication system according to 1. (2) (FIGS. 6 and 8)
(Appendix 3)
The other network delay information further includes a delay occurring in each of the other communication networks, and the own network delay information further includes a delay occurring in the own communication network,
The call control device includes delay and delay jitter that occur in each of the other communication networks, delay and delay jitter that occur in the local communication network, and delay that occurs in the second communication terminal or the first communication terminal. To calculate the total delay time, and when the total delay time is within the allowable delay time of the local communication network, the total delay jitter is set as the data buffer size, and the total delay time of the local communication network The communication system according to appendix 2, wherein when the allowable delay time is exceeded, the maximum allowable delay jitter of the communication network is set as the data buffer size. (3) (FIGS. 6 and 8)
(Appendix 4)
When the call control device receives a second communication session establishment request signal for the second communication terminal from the first communication terminal, the call control device sends the own network delay information to the call control device included in the other communication network. The communication system according to any one of appendices 1 to 3, which is notified. (4) (FIGS. 5 and 6)
(Appendix 5)
The communication system according to any one of appendices 1 to 4, wherein the other network delay information is included in the communication session establishment request signal. (Fig. 6)
(Appendix 6)
The communication system according to appendix 5, wherein the call control device establishes the communication session according to a session establishment protocol, and the communication session establishment request signal is an Invite signal. (Fig. 6)
(Appendix 7)
A first communication terminal connected to a first communication network among a plurality of communication networks that transmit and receive encoded data in units of packets, and any of the plurality of communication networks connected to the first communication network A call control device for establishing a communication session between second communication terminals connected to the communication network,
Another communication network that passes between the second communication terminal and the first communication network when a request for establishing a communication session between the second communication terminal and the first communication terminal is received. The other communication delay information relating to the delay caused by the second communication network is acquired from the other communication network, and the first communication terminal uses the other network delay information and the own network delay information relating to the delay caused in the first communication network. Obtaining a data buffer size for storing encoded data included in a packet received from the second communication terminal, and notifying the first communication terminal;
Call control device. (5) (Fig. 4)
(Appendix 8)
A communication terminal connected to a communication network that transmits and receives encoded data in units of packets,
A memory for storing encoded data included in a plurality of received packets;
The storage capacity of the memory is set to the data buffer size notified from the call control device that establishes a communication session with another communication terminal, and the capacity of the encoded data stored in the memory is set A signal processing unit that collectively decodes the accumulated encoded data;
A communication terminal. (Figure 2)
(Appendix 9)
A memory that is connected to one of at least two communication networks that transmit and receive encoded data in units of packets and that stores encoded data included in a plurality of received packets by a predetermined storage capacity, and a code stored in the memory A method for setting the predetermined storage capacity of a communication terminal having a signal processing unit that collectively decodes digitized data,
A communication session connected to a first communication network to which the communication terminal of the at least two communication networks is connected, and between the communication terminal and another communication terminal connected to one of the at least two communication networks. A call control device for establishing a communication session from the other communication terminal to the first communication network when receiving a communication session establishment request signal for the communication terminal from the other communication terminal, Obtaining other network delay information relating to a delay occurring in another communication network of at least two communication networks from the other communication network;
The call control device obtains a data buffer size from the other network delay information and own network delay information related to a delay occurring in the own communication network;
Notifying the communication terminal of the data buffer size by the call control device;
The communication terminal setting the storage capacity of the memory to the data buffer size notified from the call control device;
A storage capacity setting method for a communication terminal. (Fig. 5)

It is a schematic block diagram of the communication system by one Embodiment. It is a schematic block diagram of the IP telephone set used by one Embodiment. It is a flowchart of the storage capacity setting process of a jitter buffer. It is a schematic block diagram of the call control server used by one Embodiment. 4 is an example of a sequence for establishing a communication session in a communication system according to an embodiment. It is a flowchart of the edit process of an Invite signal. It is a flowchart of the edit process of the 180 Ringing signal transmitted to the IP telephone of a transmission origin. It is a figure which shows an example of the Invite signal transmitted to the call control server which manages another IP network. It is a figure which shows an example of the Invite signal transmitted to the IP telephone connected to the IP network which a call control server manages. It is a figure which shows an example of the 180Ringing signal transmitted to an IP telephone.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Communication system 2, 3 IP telephone 4 Telephone 10, 20 Call control server 50 Gateway 51 Router 100 Telephone switching network 200, 300 IP network 21 Voice processing part 22 Packet transmission / reception part 23 Call control part 24 Jitter buffer 31 Connection processing part 32 Recommendation Buffer size determination unit

Claims (5)

A communication system having at least two communication networks for transmitting and receiving encoded data in units of packets,
Each of the at least two communication networks is
A first communication terminal having a memory for storing encoded data included in a plurality of received packets, and a signal processing unit for collectively decoding the encoded data stored in the memory;
A call control device for establishing a communication session between the first communication terminal and a second communication terminal connected to one of the at least two communication networks;
When the call control apparatus receives a communication session establishment request signal for the first communication terminal from the second communication terminal, the call control apparatus communicates from the second communication terminal to the own communication network to which the call control apparatus belongs. Other network delay information related to a delay that occurs in another communication network among the at least two communication networks that pass between them is acquired from the other communication network, and the other network delay information and a delay that occurs in the own communication network Determining the data buffer size from its own network delay information and notifying the first communication terminal;
The first communication terminal sets the storage capacity of the memory to the data buffer size notified from the call control device, and then starts communication.
Communications system. The other network delay information includes delay jitter that occurs in each of the other communication networks, and the own network delay information includes delay jitter that occurs in the own communication network,
The call control apparatus calculates a total delay jitter by summing a delay jitter generated in each of the other communication networks and a delay jitter generated in the communication network, and sets the total delay jitter as the data buffer size. Item 12. The communication system according to Item 1. The other network delay information further includes a delay occurring in each of the other communication networks, and the own network delay information further includes a delay occurring in the own communication network,
The call control device includes delay and delay jitter that occur in each of the other communication networks, delay and delay jitter that occur in the local communication network, and delay that occurs in the second communication terminal or the first communication terminal. To calculate the total delay time, and when the total delay time is within the allowable delay time of the local communication network, the total delay jitter is set as the data buffer size, and the total delay time of the local communication network The communication system according to claim 2, wherein when the allowable delay time is exceeded, a maximum allowable delay jitter of the own communication network is set as the data buffer size.   When the call control device receives a second communication session establishment request signal for the second communication terminal from the first communication terminal, the call control device sends the own network delay information to the call control device included in the other communication network. The communication system according to any one of claims 1 to 3, which is notified. A first communication terminal connected to a first communication network among a plurality of communication networks that transmit and receive encoded data in units of packets, and any of the plurality of communication networks connected to the first communication network A call control device for establishing a communication session between second communication terminals connected to the communication network,
Another communication network that passes between the second communication terminal and the first communication network when a request for establishing a communication session between the second communication terminal and the first communication terminal is received. The other communication delay information relating to the delay caused by the second communication network is acquired from the other communication network, and the first communication terminal uses the other network delay information and the own network delay information relating to the delay caused in the first communication network. Obtaining a data buffer size for storing encoded data included in a packet received from the second communication terminal, and notifying the first communication terminal;
Call control device.

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