JP2007116368A - Voice ip packet switch - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To warrant the continuity of voice IP packets to an opposite terminal even in the case that a sender terminal is switched. <P>SOLUTION: The voice IP packet switch includes: a plurality of voice IP packet reception decomposing means that receive an arrived voice IP packet to extract a voice payload; a voice payload switch means that switches the voice payload extracted by any of the voice IP packet reception decomposing means in response to connection information between a sender side terminal and a receiver side terminal used for the communication of the voice IP packet related to the voice payload; and a plurality of voice IP packet buildup and transmission means that attaches a packet header to the voice payload received from the voice payload switch means to build up the voice IP packet and transmits the voice IP packet to the receiver side IP phone terminal. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to a voice IP packet switching apparatus, and can be applied to, for example, an IP-PBX apparatus.

  Various telephone networks exist, and voice packet switching apparatuses corresponding to a plurality of telephone networks exist (see Patent Document 1). For example, an IP-PBX device is also a device that exchanges voice data between an extension network and an outside network.

  FIG. 2 is a block diagram showing a processing configuration of an audio signal from an extension line to an outside line in a conventional IP-PBX apparatus. In the following, in the conventional IP-PBX device, the process of inputting voice from the extension IP telephone 10, the process of inputting voice from the analog telephone 12, the process of outputting voice to the outside IP telephone 11, and the analog of the outside line A process for sending voice to the telephone 13 will be described.

(P1) Processing for Inputting Voice from Extension IP Phone 10 Voice packets (IP packets) from the extension IP phone 10 are sent to the voice packet receiver 20. The voice packet receiving unit 20 refers to the connection information in the connection information table 30 registered by the call control unit 40 and determines where to output the received voice packet. If the output destination is the external line IP telephone set 11, the received voice packet is sent to the voice packet transmission unit 21 as it is, and if the output line is the external line analog telephone set 13, the voice data part is extracted from the voice packet, and the analog voice transmission unit Send to 23.

(P2) Processing for Inputting Voice from Extension Analog Telephone 12 A voice signal from the extension analog telephone 12 is sent to the analog voice receiver 22. The analog voice receiving unit 22 performs A / D conversion processing and processing to cut out and packetize at regular intervals to create voice data. Further, the connection information of the connection information table 30 registered by the call control unit 40 is referred to determine where to output the voice packet (IP packet). If the output destination is an external IP phone 11, an appropriate packet header is added, a voice packet is created, and then sent to the voice packet transmitter 21. If the output destination is an analog telephone 13 on the outside line, the voice data is sent to the analog voice transmission unit 23 as it is without adding a packet header.

(P3) Processing for outputting voice to IP telephone 11 on the outside line Voice packets (IP network) received by voice packet transmitting unit 21 are sent to a network (IP network) using network processing such as an OS, and IP It is output to the telephone 11.

(P4) Processing for sending voice to external line analog telephone set 13 The voice data received by the analog voice transmission unit 23 is arranged in time series, D / A converted, sent to the PSTN network, and sent to the analog telephone 13 Is output.
JP 2003-60711 A

  However, in the conventional IP-PBX device (voice IP packet switching device), there is a possibility that a problem may occur when performing an additional service on the telephone such as call transfer.

  FIG. 3 is an explanatory diagram of such a problem. FIG. 3A shows the flow of a voice signal from the IP telephone 10 to the IP telephone 11 before the call transfer, and FIG. 3B shows the voice signal from the analog telephone 12 to the IP telephone 11 after the call transfer. The flow is shown. That is, FIG. 3 shows a case of a call transfer service in which the call terminal is switched from the IP phone 10 to the analog phone 12 while the call is maintained while the call is being performed between the IP phone 10 and the IP phone 11. Show.

  When such a call transfer is performed, before the call transfer, the IP telephone 10 performs a voice packet header addition operation, and the voice packet from the IP telephone 10 passes through the voice packet receiving unit 20 and the voice packet transmitting unit 21. To the opposite IP telephone 11. On the other hand, after the call transfer, the analog voice receiving unit 22 adds a header to the voice data sent from the analog telephone 12 to complete the voice packet, and the voice packet is opposed via the voice packet transmitting unit 21. Is sent to the IP telephone 11.

  As described above, since the header addition operation is executed at different places before and after the call transfer, the header is no longer relevant before and after the call transfer. Specifically, discontinuity occurs in information such as a sequence number and a time stamp included in the header. Further, when viewed from the IP-PBX device (voice IP packet switching device), since the IP telephone 10 that performs the header addition operation is an external device, it is difficult to achieve synchronization. In the IP telephone 11 on the receiving side, due to the discontinuity of the header, it is determined that the header is abnormal in the voice packet, which may cause a problem such as voice interruption.

  Therefore, there is a demand for a voice IP packet switching apparatus that can guarantee the continuity of voice IP packets to the opposite terminal even when the source terminal is switched.

  The present invention provides a voice IP packet switching apparatus for exchanging voice IP packets, receiving a voice IP packet that has arrived and extracting a voice payload, and a plurality of voice IP packet reception and decomposition means, and any one of the above voice IP packet reception and decomposition A voice payload exchanging means for exchanging the voice payload extracted by the means according to connection information between the transmitting side terminal and the receiving side terminal used for communication of the voice IP packet related to the voice payload; A plurality of voice IP packet construction and transmission means for adding a packet header to construct a voice IP packet and transmitting it to the receiving IP telephone terminal when a voice payload is given from the means; It is characterized by.

  According to the present invention, even if a voice IP packet is inputted, it is exchanged back to the voice payload, and a packet header is added to the voice payload after the exchange to construct and send the voice IP packet. Even if the original terminal is switched, the continuity of the output voice IP packet can be guaranteed.

(A) Embodiment Hereinafter, an embodiment in which the voice IP packet switching apparatus of the present invention is applied to an IP-PBX apparatus will be described with reference to the drawings.

(A-1) Configuration of Embodiment FIG. 1 is a block diagram illustrating a processing configuration of an audio signal from an extension line to an outside line in the IP-PBX apparatus according to the embodiment. In FIG. 1, the signaling processing configuration is omitted except for the call control unit.

  In FIG. 1, an IP-PBX device 1 according to the embodiment has a plurality (only one is shown in FIG. 1) of IP telephones 10 and a plurality (only one is shown in FIG. 1) as extension terminals. The analog telephone 12 is accommodated, and the IP telephone 11 and the analog telephone 13 are used as external line terminals capable of communication (in other words, the external network capable of communication is an IP network and a PSTN network).

  The IP telephone 10 as an extension terminal may be a fixed telephone, a mobile phone via an access point device, an IP telephone as a hard phone, or an IP telephone as a soft phone. good. Alternatively, the IP-PBX device 1 and the IP telephone 10 may be directly connected via a cable, and the IP-PBX apparatus 1 accommodates the IP telephone 10 via a LAN or the like. It may be. That is, the IP telephone 10 may be a telephone that exchanges audio signals with the IP-PBX device 1 using IP packets (voice IP packets).

  The analog telephone 12 as an extension terminal may be a fixed telephone or a mobile telephone via an access point device. The IP-PBX device 1 and the analog telephone 12 may be directly connected via a cable, and the IP-PBX device 1 accommodates the analog telephone 12 via a LAN or the like. It may be. In the following description, the analog telephone 12 is described as transmitting and receiving an analog voice signal, but digital voice data that is not packetized may be transferred to and from the IP-PBX device 1 (FIG. 1). Is written as an audio payload). The analog telephone 12 may handle a conventional narrowband audio signal, or may handle a broadband audio signal that has been proposed in recent years.

  As described above, the IP-PBX device 1 according to the embodiment accommodates the IP telephone 10 that transmits / receives voice IP packets and the analog telephone 12 that transmits / receives non-packet voice payloads.

  In FIG. 1, the IP-PBX device 1 of the embodiment functionally includes a voice packet receiving unit 20, a voice packet transmitting unit 21, an analog voice receiving unit 22, an analog voice transmitting unit 23, a voice data extracting unit 24, and a packet. A header adding unit 25, a voice payload exchanging unit 26, a connection information registration table 30, a header information storage table 31, and a call control unit 40 are included.

  There are a plurality of voice packet receiving units 20, and each of the voice packet receiving units 20 receives a voice IP packet sent from the corresponding extension IP telephone 10 (which may be fixed or for each call), and To the voice data extraction unit 24.

  There are a plurality of voice packet transmission units 21, and the voice packet transmission unit 21 transmits the voice IP packet given from the corresponding packet header addition unit 25 to the IP network, and hence the IP telephone 11 on the outside line.

  There are a plurality of analog voice receivers 22, and each analog voice receiver 22 receives an analog voice signal given from the corresponding analog telephone 12, A / D-converts the analog voice signal, and then at regular intervals. A process of cutting out is performed, and the processed audio signal (audio payload) is supplied to the audio payload exchange unit 26 together with transmission source information representing the analog telephone 12. That is, a voice payload that can be inserted into a voice IP packet is created and given to the voice payload exchange unit 26 together with transmission source information representing the analog telephone 12. Here, the analog voice receiving unit 22 and the extension analog telephone 12 have a one-to-one correspondence, and the analog voice receiving unit 22 can add transmission source information. Note that the analog voice receiving unit 22 may perform an encoding process according to a predetermined encoding method after A / D conversion, and then perform a process of cutting out at regular intervals. If an A / D converted audio signal is input, only the subsequent processing described above may be executed.

  As described above, the analog voice receiving unit 22 of the embodiment outputs a voice payload to which transmission source information is added, and does not complete a conventional voice IP packet.

  There are a plurality of analog voice transmission units 23, and each of the analog voice transmission units 23 performs a D / A conversion after sequentially converting audio payloads not including a header given from the audio payload exchange unit 26 into sequential audio data. The analog audio signal obtained is output to the analog telephone 13 on the outside line via the PSTN network. Note that a decoding process according to a predetermined encoding method may be performed before the D / A conversion. Also, depending on the type of PSTN network, sequential audio data may be output as it is.

  Since the input signal is not a voice IP packet but a voice payload, the analog voice transmitter 23 of the embodiment does not remove the header unlike the conventional one.

  The voice data extracting unit 24 extracts a voice data portion (voice payload) from the voice IP packet received by the corresponding voice packet receiving unit 20, and supplies the voice data exchanging unit 26 together with the transmission source information of the voice IP packet to the voice payload exchanging unit 26. It is. The transmission source information may be obtained by extracting from the header, or may be obtained from the port number received by the voice packet receiving unit 20 or the like. Alternatively, the set of the voice packet receiving unit 20 and the voice data extracting unit 24 may be associated with the extension IP telephone 10 on a one-to-one basis so that transmission source information can be added.

  The packet header adding unit 25 completes the voice IP packet by adding the packet header to the voice payload not including the header given from the voice payload exchanging unit 26 while referring to and operating the stored contents of the header information table 31. To the corresponding voice packet transmitter 21.

  The audio payload exchanging unit 26 refers to the audio payload added with the transmission source information given from the audio data extraction unit 24 or the analog audio reception unit 22 while referring to the stored contents of the connection information registration table 30. Are exchanged and output to the analog voice transmitting unit 23 or the packet header adding unit 25 associated with the opposite terminal during a call. The output data from the audio payload exchange unit 26 is only the audio payload from which the transmission source information is excluded. The transmission source information may be output to the analog voice transmission unit 23 or the packet header addition unit 25, and the removal of the transmission source information may be a function of the analog voice transmission unit 23 or the packet header addition unit 25.

  The connection information registration table 30 stores group information (and group information between extension terminals) and the like of extension terminals and external terminals that are in a call state (the communication path is established) under the control of the call control unit 40. It is what. In other words, the connection information registration table 30 stores information for exchange necessary for the voice payload exchange unit 26 to perform exchange processing.

  FIG. 4 is an explanatory diagram of an example of the configuration of the connection information registration table 30. In the configuration example shown in FIG. 4, first connection destination information that directly or indirectly represents a transmission source terminal in the transmission direction among the two terminals in a call state in each row, and the transmission destination terminal The second connection destination information expressed directly or indirectly is described in association with it. Here, information is set in each row when a speech path is established. Note that the connection destination information is replaced with the identification information of the terminal, and the constituent elements of the IP-PBX device 1 (for example, the analog voice receiving unit 22, the voice data extracting unit 24, and the analog voice transmitting unit 23) interposed on the communication path. The identification information of the packet header adding unit 25) may be used. Further, the connection information may be a combination of the input port number and the output port number of the voice payload exchange unit 26. Furthermore, the second connection information registration in which the first connection information registration table having the configuration shown in FIG. 4, the second connection destination information, and the identification information of the analog voice transmission unit 23 or the packet header addition unit 25 are associated with each other. The connection information registration table 30 may be configured by the table. Furthermore, the connection information registration table 30 may be configured regardless of the audio signal transmission direction.

  The header information storage table 31 stores header information (all or a part thereof) for the packet header adding unit 25 to add to the audio payload.

  FIG. 5 is an explanatory diagram of an example of the configuration of the header information storage table 31. In the configuration example shown in FIG. 5, each row corresponds to each of the packet header adding unit 25, and each row includes identification information of the destination terminal (may be an IP address or a telephone number), SSRC ( Synchronization source identifier), sequence number, time stamp, and the like. Here, SSRC is a value for distinguishing voice IC packets for each call, and one fixed value is continuously used in one call (session). The sequence number is a value indicating the number of the transmitted packet, and is incremented by 1 every time one packet is transmitted. The time stamp represents a time when a packet is transmitted, and a value to be increased is determined according to the packet transmission cycle, the type of speech encoding, and the like. The setting (initial setting) of each row in the header information storage table 31 is executed, for example, by the call control unit 40. At this time, the identification information of the destination terminal and SSRC are written, and the sequence number, time stamp, etc. are initialized. The subsequent operations such as sequence number and time stamp are executed by the packet header adding unit 25. In the case of the configuration example of FIG. 5, transmission source information is not included, but each packet header adding unit 25 has identification information unique to the packet header adding unit 25 (an IP address or a telephone number may be used). May be given as transmission source information.

  The call control unit 40 performs call control such as establishment of a communication path (session), disconnection, and additional services such as call transfer. For example, the call control unit 40 can accommodate the extension IP telephone 10 accommodated therein, the IP network of the outside line, etc. Control is performed using a call control protocol such as H.323, and the accommodated extension analog telephone 12 controls an external PSTN network using a call control protocol for the analog network.

(A-2) Operation of Embodiment Next, the operation at the time of a call of the IP-PBX device 1 of the embodiment is a process of inputting voice from the extension IP telephone 10, a process of inputting voice from the extension analog telephone 12, The input voice exchange process, the process of outputting voice to the external line IP telephone set 11, and the process of outputting voice to the external line analog telephone set 13 will be described separately.

  Note that the transmission / reception of signaling signals and the call control processing itself are substantially the same as those in the prior art, and a description thereof will be omitted. However, when a new communication path (session) is established, the call control unit 40 registers the connection information related to the established communication path in the connection information registration table 30, and if the outside line side is the IP telephone 11. The header information storage table 31 is provided with a row relating to the established call path, and the connection destination information and SSRC are set.

(S1) Processing when inputting voice from the IP telephone 10 The voice IP packet sent from the IP telephone 10 is received by the voice packet receiving unit 20 and given to the voice data extracting unit 24. The voice data extraction unit 24 extracts a voice data portion (voice payload) from the received voice IP packet, adds transmission source information that is received from the IP telephone 10 side to the voice payload exchange unit 26. give.

(S2) Processing when Inputting Voice from Analog Telephone 12 Voice (analog voice signal) from the analog telephone 12 is received by the analog voice receiver 22. The analog voice receiving unit 22 cuts out voice at an A / D conversion process or at regular intervals, and creates a voice data portion (voice payload) in the voice IP packet. Then, the analog voice receiving unit 22 adds the transmission source information that it has been received from the analog telephone side 12 to this voice payload and gives it to the voice payload exchanging unit 26.

(S3) Exchange processing of input voice When the voice payload exchanging unit 26 receives the voice payload to which the source information is added from the voice data extracting unit 24 or the analog voice receiving unit 21, the added source The connection information registration table 30 having the information as the first connection destination information is searched for, and the packet header addition unit 25 or the analog voice transmission unit 23 defined by the second connection destination information in the row receives the information. Output the audio payload.

(S4) Processing for outputting voice to IP telephone 11 on the outside line When a voice payload is given from voice payload exchanging section 26, packet header adding section 25 converts necessary header information (all or part of header information) into header information. Obtained from the storage table 31, adds the obtained header information to the voice payload, completes the voice IP packet, and gives it to the voice packet transmitter 21. The voice packet transmitter 21 sends the voice IP packet to the IP network and accordingly , To the corresponding IP telephone 11 of the outside line.

  Further, the packet header adding unit 25 updates the information related to the transmission destination in the header information storage table 31 to the header information for the next transmission. For example, the sequence number is incremented by 1 or the time stamp is updated to match the next transmission time.

(S5) Processing for outputting voice to the analog telephone 13 on the outside line When the voice payload (voice data) is given from the voice payload exchanging section 26, the analog voice transmitter 23 continuously receives the received voice data in time series. After arranging the voice data, D / A conversion is performed to convert the voice data into an analog voice signal, which is output to the corresponding analog telephone 13 via the PSTN network.

  The voice from the extension IP phone 10 is transmitted to the outside IP phone 11 through the above-described processing of s1-s3-s4 (see FIG. 6A). The voice from the extension analog telephone 12 is transmitted to the external IP telephone 11 through the above-described processing of s2-s3-s4 (see FIG. 6B). Further, the voice from the extension IP telephone 10 is transmitted to the analog telephone 13 of the outside line through the processing of s1-s3-s5 described above. Furthermore, the voice from the extension line analog telephone 12 is transmitted to the outside line analog telephone 13 through the above-described processing of s2-s3-s5.

  Next, a brief description will be given of changes in the voice path during call transfer for switching the call state between the extension IP telephone 10 and the external line IP telephone 11 to the call state between the extension analog telephone 120 and the external line IP telephone 11. .

  Before the call transfer, the voice from the extension IP telephone 10 passes through the voice packet receiving unit 20 -the voice data extracting unit 24 -the voice payload exchanging unit 26 -the packet header adding unit 25 -the voice packet transmitting unit 21, It is sent to the IP telephone 11 of the opposite outside line. Here, the header of the voice IP packet output from the IP-PBX device 1 is added by the header adding unit 25.

  The call transfer procedure may be arbitrary, but an example is as follows. For example, when the extension side user wishes to transfer a call, the user presses the hold button of the extension IP telephone 10 to go on-hook. At this time, the call control unit 40 blinks the external line button in all the internal terminals that have been associated with the external line IP telephone 11 until now. Thereafter, when the extension-side user presses the blinking outside line button after off-hooking the extension line analog telephone 12, the call control unit 40 calls the other party's telephone partner with the outside line IP telephone 11 to the extension line analog telephone 12. Switch. At this time, for example, the call control unit 40 obtains the first connection destination information in the row where the second connection destination information is related to the IP telephone 11 in the connection information registration table 30 from the identification information of the IP telephone 10 and the analog telephone 12. The process of rewriting to the identification information is executed.

  After such a call transfer, the voice from the extension analog telephone 12 passes through the analog voice receiving unit 22 -voice payload exchanging unit 26 -packet header adding unit 25 -voice packet transmitting unit 21 and the opposite external line. To the IP telephone 11. Again, the header of the voice IP packet output from the IP-PBX device 1 is added by the header adding unit 25.

  That is, the header of the IP voice packet sent to the opposite external IP telephone 11 is managed by the same packet header addition unit 25 in the same row of the header information storage table 31 before and after call transfer. It is added based on the header information.

(A-3) Effect of Embodiment As described above, according to the above-described embodiment, the voice data main body without a header can be used regardless of whether the input voice from the extension terminal is an IP packet or another format. When the external line of the output destination is an IP telephone set, a header is added to form a voice IP packet. Therefore, even if the extension terminal is switched due to a call transfer service or the like, the external line IP It is possible to guarantee continuity in the header of the voice IP packet to the telephone, and it is possible to prevent the external IP telephone from malfunctioning.

(B) Other Embodiments In the above embodiment, the analog audio receiving unit 22 and the audio data extracting unit 24 add the transmission source information and give the audio payload to the audio payload exchanging unit 26. If the payload exchange unit 26 is configured to recognize the transmission source from an input port or the like, the analog audio reception unit 22 or the audio data extraction unit 24 may simply provide the audio payload to the audio payload exchange unit 26. good.

  In the above embodiment, the present invention is applied in the direction from the inner line side to the outer line side. However, the present invention may be applied to the reverse direction from the outer line side to the inner line side. In addition, regarding the reverse direction, the same configuration as the conventional one (see FIG. 2) may be applied.

  Further, in the above embodiment, the present invention is applied in the direction from the extension side to the outside line side. However, the voice path from the extension side to the extension side also includes a voice packet reception unit, a voice data extraction unit, a voice The path may be configured by components such as a payload exchange unit, a packet header addition unit, and a voice packet transmission unit. In this case, for example, the present invention can also be applied to a case where an extension IP telephone is accommodated in the IP-PBX apparatus via a public IP network.

  Furthermore, in the above-described embodiment, the present invention is applied to an IP-PBX device that accommodates IP telephones and analog telephones as extension terminals. However, the present invention is applicable to an IP-PBX apparatus that accommodates only IP telephones. Can be applied.

  In the above embodiment, the present invention is applied to an IP-PBX device. However, other than a voice IP packet switching device for a station, an IP telephone accommodating terminal for home use (for example, a VoIP router), etc. The present invention can be applied to other voice IP packet switching apparatuses.

  Although the present invention is intended for transmission on a voice channel, it goes without saying that a transmission signal on a voice channel is not limited to a voice signal. However, in the expression of the scope of claims, the expression “speech” is used, although other types of signals are included.

It is a block diagram which shows the processing structure of the audio | voice signal from the internal line to an external line in the IP-PBX apparatus of embodiment. It is a block diagram which shows the processing structure of the audio | voice signal from the internal line to an external line in the conventional IP-PBX apparatus. It is explanatory drawing of the subject of the conventional IP-PBX apparatus shown in FIG. It is explanatory drawing which shows the structural example of the connection information registration table of embodiment. It is explanatory drawing which shows the structural example of the header information storage table of embodiment. It is explanatory drawing which shows the path | route of the audio | voice signal before and after the call transfer of embodiment.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... IP-PBX apparatus, 10 ... IP telephone, 12 ... Analog telephone, 20 ... Voice packet receiver, 21 ... Voice packet transmitter, 22 ... Analog voice receiver, 23 ... Analog voice transmitter, 24 ... Voice data extraction 25: Packet header adding unit, 26: Voice payload exchange unit, 30 ... Connection information registration table, 31 ... Header information storage table, 40 ... Call control unit.

Claims (3)

In a voice IP packet switching apparatus that exchanges voice IP packets,
A plurality of voice IP packet receiving and decomposing means for receiving an incoming voice IP packet and extracting a voice payload;
The voice to be exchanged according to the connection information between the transmitting side terminal and the receiving side terminal that is used for communication of the voice IP packet related to the voice payload, with the voice payload extracted by any of the above voice IP packet receiving and decomposing means Payload exchange means;
When a voice payload is given from the voice payload exchanging means, a plurality of voice IP packet constructing / transmitting means for adding a packet header, constructing a voice IP packet, and transmitting the voice IP packet to the receiving IP telephone terminal A voice IP packet switching apparatus characterized by comprising: One or a plurality of analog voice receiving means for forming a voice payload that becomes a voice IP packet if a packet header is added from an incoming analog voice signal and inputting the voice payload to the voice payload exchange means;
One or a plurality of analog voice transmission means for converting the received voice payload sequence into an analog voice signal and sending it to the analog telephone terminal on the receiving side when the voice payload is given from the voice payload exchange means; The voice IP packet switching apparatus according to claim 1, further comprising: Each of the voice IP packet receiving / decomposing means or the analog voice receiving means provides the voice payload exchange means with source information defining a source terminal together with the voice payload,
The voice IP packet switching device according to claim 1 or 2, wherein the voice payload switching means recognizes the corresponding connection information based on given transmission source information.

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