JP6780978B2 - Information record control device and information record control method - Google Patents

Description

The present invention relates to an information recording control device and an information recording control method.

Conventionally, a call recording device for recording a call in a Voice over IP (VoIP) environment has been developed. In a VoIP environment, voice is carried by RTP (Real-time Transport Protocol) packets. Therefore, in order to record a call in a VoIP environment, a mechanism for recording from an RTP packet is required.

For example, in a call recording system connected to a mirror port of a network switch, an RTP packet flowing through an IP network is acquired from the mirror port, an uplink and downlink RTP packets are judged as one call, recorded, and a recorded file is recorded. Generated and stored (see, for example, Patent Document 1).

To obtain the upstream and downstream RTP packet streams that make up a call from an RTP packet, for example, a SIP message exchanged between the calling terminal and the incoming terminal via a SIP server. From the call control information of, the IP addresses of the calling side terminal and the calling side terminal are acquired, and the IP packets using these IP addresses as the destination addresses are regarded as uplink and downlink RTP packets and acquired. There is.

Japanese Unexamined Patent Publication No. 2013-187733

The service of call recording is required to prevent voice loss due to the nature of leaving evidence of commercial transactions and customer support, for example.

In the conventional call recording device as described above, the call recording is started and ended based on the call control information of the SIP message. However, if there is a change in the control process of the SIP server, the call recording device cannot follow the change, so that the call control information cannot be used and the call cannot be recorded, resulting in omission.

Further, a plurality of extension networks may be constructed in one network, and different SIP servers may be operated in each extension network. In this case, even if the call recording device can follow one SIP server, it may not be able to follow the other SIP server. In such a situation, in some calls, the call control information cannot be used and the call cannot be recorded, so that the call is missed.

In addition, there is a type of call recording device that acquires call control information in cooperation with a private telephone exchange (hereinafter, also referred to as "PBX"). Even with this type, new functions and services have been added to the PBX, and since the call recording device cannot keep up with the changes, it may not be possible to link with the PBX and call recording may not be possible, resulting in omissions.

Further, the call recording device acquires not only the start / end of call recording but also additional information such as a caller ID from the call control information based on the call control information. For this reason, it is desired that the call recording device mainly performs call recording in cooperation with SIP messages or PBX, and can also record calls that are not covered only by cooperation with these.

Such a problem is also the same when recording video data transmitted and received between videophones other than the above-mentioned call recording.

The present invention has been made in view of such a point, and an object of the present invention is to provide an information recording control device and an information recording control method capable of realizing comprehensive recording regardless of whether or not call control information can be used. ..

One aspect of the information recording control device of the present invention is a first session processing unit that determines a packet stream corresponding to call control information acquired from a call control packet as a recording target, and a packet stream that does not correspond to the call control information. A second session processing unit that determines to be a recording target, a packet stream processing unit that collects the packet stream of the recording target determined by the first session processing unit and the second session processing unit, and a packet stream processing unit collects the packet stream. A main information recording unit that generates main information from a plurality of main information packets constituting the packet stream and causes a recording device to record the main information, and is known for each of the plurality of received packets flowing through the network. It is determined whether or not the packet stream is the main information packet constituting the packet stream, and the received packet is processed by any one of the second session processing unit and the packet stream processing unit based on the determination result. It is characterized by.

With this configuration, it is determined whether or not each of the plurality of received packets flowing through the network is a known main information packet constituting the packet stream to be recorded, and based on the determination result, the received packet is subjected to the first. Since any one of the two-session processing unit and the packet stream processing unit is used for processing, comprehensive recording can be realized regardless of whether or not the call control information can be used.

In one aspect of the information recording control apparatus of the present invention, the call control information, it is preferable that information about the node obtained from the call control packet.

In one aspect of the information recording control device of the present invention, when the received packet is the call control packet, the received packet is output to the first session processing unit, and the received packet is a known packet to be recorded. If it is not a stream, the received packet is output to the second session processing unit, and if the received packet is a known packet stream to be recorded, the received packet is output to the packet stream processing unit. It is preferable to further include a portion.

In one aspect of the information recording control device of the present invention, it is preferable that the call control information is information about a node acquired from a private telephone exchange.

In one aspect of the information recording control device of the present invention, it is determined whether or not the received packet is a known call control packet based on the information about the node acquired from the private telephone exchange, and the received packet is known. If it is not the packet stream to be recorded, the received packet is output to the second session processing unit, and if the received packet is a known packet stream to be recorded, the received packet is sent to the packet stream processing unit. It is preferable to further include a received packet distribution unit for output.

In one aspect of the information recording control device of the present invention, it is preferable that the main information packet is an RTP packet.

In one aspect of the information recording control device of the present invention, it is preferable that the main information is at least one selected from voice, image, video and text.

One aspect of the information recording control method of the present invention comprises a first determination step of determining a packet stream corresponding to call control information as a recording target, and a received packet flowing through the network constitutes a known packet stream of the recording target. The second determination step of determining whether or not the packet is the main information packet, the third determination process of determining the received packet as the recording target of the packet stream that does not correspond to the call control information, and the packet stream. It is provided with a step of collecting a plurality of main information packets constituting the above, generating main information from the plurality of main information packets, and attaching to any one of packet stream processing to be recorded by a recording device. It is a feature.

With this configuration, it is determined whether or not each of the plurality of received packets flowing through the network is a known main information packet constituting the packet stream to be recorded, and based on the determination result, the received packet is called. A third determination process for determining a packet stream that does not correspond to the control information as a recording target, and a recording device that collects a plurality of main information packets constituting the packet stream and generates main information from the plurality of main information packets. Since it is attached to any one of the packet stream processing to be recorded in, comprehensive recording can be realized regardless of whether the call control information can be used or not.

According to the present invention, it is possible to provide an information recording control device and an information recording control method capable of realizing comprehensive recording regardless of whether or not call control information is available.

It is a schematic diagram which shows the whole structure of the call recording system including the call recording device which concerns on 1st Embodiment. It is a schematic diagram which shows the 1st extension network and the 2nd extension network in the call recording system shown in FIG. It is a block diagram which shows the hardware of the server which comprises the call recording apparatus which concerns on 1st Embodiment. It is a functional block diagram which shows the call recording apparatus which concerns on 1st Embodiment. It is a flowchart which shows the processing in the received packet distribution part in the call recording apparatus which concerns on 1st Embodiment. It is a flowchart which shows the process in the RTP session part in the call recording apparatus which concerns on 1st Embodiment. It is a sequence diagram which shows the call recording procedure in the call recording apparatus which concerns on 1st Embodiment. It is a sequence diagram which shows the call recording procedure in the call recording apparatus which concerns on 1st Embodiment. It is explanatory drawing which shows the RTP session processing with respect to the RTP packet which constitutes both the up and down packet streams shown in FIG. It is a schematic diagram which shows the whole structure of the call recording system including the call recording device which concerns on 2nd Embodiment. It is a schematic diagram which shows the 1st extension network and the 2nd extension network in the call recording system shown in FIG. It is a functional block diagram which shows the call recording apparatus which concerns on 2nd Embodiment. It is a flowchart which shows the processing in the received packet distribution part in the call recording apparatus which concerns on 2nd Embodiment. It is a sequence diagram which shows the call recording procedure in the call recording apparatus which concerns on 2nd Embodiment. It is a sequence diagram which shows the call recording procedure in the call recording apparatus which concerns on 2nd Embodiment.

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

(Overview)
The information recording control device according to the present embodiment has a first session processing unit that determines a packet stream corresponding to call control information as a recording target, and a second session processing unit that determines a packet stream that does not correspond to the call control information as a recording target. It constitutes a session processing unit, a packet stream processing unit that collects the packet stream to be recorded determined by the first session processing unit and the second session processing unit, and the packet stream collected by the packet stream processing unit. Whether or not the received packet flowing through the network is a main information packet constituting the known packet stream to be recorded, including a main information recording unit that generates main information from a plurality of main information packets and causes the recording device to record the main information. Based on the determination result, the received packet is processed by any one of the second session processing unit and the packet stream processing unit.

Further, in the information recording control method according to the present embodiment, the first determination step of determining the packet stream corresponding to the call control information as the recording target, and the received packet flowing through the network are known packet streams of the recording target. A second determination step for determining whether or not the packet is a constituent main information packet, a third determination process for determining the received packet as a recording target for a packet stream that does not correspond to the call control information, and the above. The present invention includes a step of collecting a plurality of main information packets constituting a packet stream, generating main information from the plurality of main information packets, and subjecting the packet stream processing to one of the packet stream processes of recording in a recording device. ..

In the information recording control device according to the present embodiment, the first session processing unit determines that the packet stream corresponding to the call control information is the recording target.

Here, the call control information is information about a node used to establish a session between two or more nodes. The information about the node includes, for example, the IP address assigned to the node and the telephone number (external line number and extension number) assigned to the node, but is not particularly limited.

The call control information can be obtained from, for example, a call control packet containing the call control information. In this case, the call control information is specifically referred to as SIP (Session Initiation Protocol) and H.A. It is used in the 323 protocol and the like, but is not particularly limited. Hereinafter, SIP will be described as an example.

In SIP, the SIP message, which is the call control information, is in text format and is divided into a start line, a header part, and a body part. Call control information such as compression format is described.

The call control packet is an IP packet containing the above-mentioned call control information.

In the case of SIP, for example, the calling party address can be obtained from the body portion of the INVITE request. On the other hand, the called party address can be obtained from the body portion of the 200 OK response. However, the calling side address and the called side address may be acquired from other call control packets.

The first session processing unit analyzes, for example, the call control information included in the call control packet, and acquires the calling party address which is the IP address of the calling party node and the called party address which is the IP address of the called party. ..

The call control information can also be obtained from a private telephone exchange (PBX). In this case, the first session processing unit uses the API (application program interface) provided by the server constituting the PBX to acquire information about the node from the PBX.

The first session processing unit determines that the packet stream corresponding to the call control information is to be recorded based on the call control packet or the call control information acquired from the PBX.

Here, the packet stream is a data stream for transmitting main information transmitted and received between a calling side node and a calling side node, and is composed of a plurality of main information packets.

In the first session processing unit, for example, based on the calling side address (A) and the called side address (B) acquired from the call control information, a packet stream (A) having the calling side address (A) as the destination address ( The packet stream (downstream packet stream) having the incoming packet stream) and the called party address (B) as the destination address is determined to be the recording target.

The first session processing unit outputs information about the packet stream determined as a recording target based on the determination result to the packet stream processing unit.

On the other hand, the second session processing unit determines that the packet stream that does not correspond to the call control information is the recording target.

The packet stream that does not correspond to the call control information is a packet stream that cannot be identified based on the above-mentioned call control packet or the call control information acquired from the PBX.

The second session processing unit analyzes, for example, a plurality of main information packets having the same destination address without using the call control information, and determines the packet stream as a recording target. The method of determination is not particularly limited, but is, for example, the data structure of the main information packet, the identity of the synchronous source (SSRC) identifier of the main information packet, the continuity of the sequence number of the main information packet, and the main information packet. Use at least one of the judgment conditions such as the number.

The second session processing unit outputs information about the packet stream determined as a recording target based on the determination result to the packet stream processing unit.

The packet stream processing unit holds information about the packet stream to be recorded input from the first session processing unit and the second session processing unit. The information regarding the packet stream is, for example, a source address (srcaddr) and a destination address (dstatdr) of the main information packet constituting the packet stream, but is not particularly limited.

The packet stream processing unit collects the main information packets constituting the packet stream to be recorded by using the information about the packet stream to be recorded, and inputs the main information packets to the main information recording unit.

The main information recording unit generates main information from a plurality of main information packets input from the packet stream processing unit, and stores the main information in the storage device.

Here, the main information is data transmitted and received between the nodes. For example, the main information is multimedia data, more specifically at least one selected from audio, image, video and text. In an IP telephone using VoIP, the main information is audio and / or video. Hereinafter, the case where the main information is voice will be described as an example.

The main information packet is an IP packet containing a part of the main information. When the main information is voice, for example, RTP is used as a protocol for transmitting and receiving the main information.

RTP is, for example, based on, but not limited to, UDP. The RTP packet is composed of an IP header, a UDP header, an RTP header, and a data unit, and the divided voice data is stored in the data unit. The RTP contains a sequence number and a time stamp. In RTP, in order to correctly transmit the voice data transmitted by the transmitting node, the receiving node sorts the packets in the correct order according to the sequence number. In addition, a voice call is realized by determining the timing of voice reproduction based on the time stamp information on the receiving node. A plurality of main information packets arranged in order in this way are called a packet stream.

As represented by RTP, the present invention includes data communication including a destination address and a source address, multimedia data as main information, and information indicating the order of packets by a sequence number or the like. It can be applied to the protocol.

In the case of VoIP described here, the call control packet conforms to SIP and the main information packet conforms to RTP, but the call control packet and the main information packet may conform to the same protocol. An example of such a case is WebRTC (Web Real-Time Communication).

In the case of RTP, as described above, the main information recording unit sorts a plurality of RTP packets constituting the packet stream in the correct order according to the sequence number, determines the timing of voice reproduction based on the time stamp information, and goes up and down. Generate call voice data from both packet streams. In addition, the main information recording unit converts the generated call voice data into a voice file.

The storage device for storing the main information may be built in the information recording control device or may be connected from the outside. The storage device may be connected to the information recording control device by a serial cable, USB, or the like, or may be connected to the information recording control device by a network such as a wired LAN or a wireless LAN. The storage device is a hard disk drive, a solid state drive (SSD), a magnetic disk drive, an optical disk drive, or the like, but is not particularly limited.

The storage device is more preferably an information recording server connected to the information recording control device via a network. The information recording server is configured to be able to search for and reproduce the stored main information from a personal computer (PC) connected via a network.

In the information recording control device according to the present embodiment, it is determined whether or not the received packet constitutes a known packet stream to be recorded, and the received packet is transmitted to the second session processing unit and the packet based on the determination result. Let any one of the stream processing units process.

The known packet stream to be recorded is a packet stream in which information about the packet stream to be recorded is held in the packet stream processing unit.

When the call control information is information about a node acquired from the call control packet, if the received packet is a call control packet, the received packet is processed by the first session processing unit, and the received packet is a known packet stream to be recorded. If not, it is preferable to have the second session processing unit process the received packet, and if the received packet is a known packet stream to be recorded, have the packet stream processing unit process the received packet.

When the call control information is information about a node acquired from PBX, it is determined whether or not the received packet is a known packet stream based on the information about the node acquired from PBX, and if it is not a known packet stream to be recorded, it is received. It is preferable to have the second session processing unit process the packet, and when the received packet is a known packet stream to be recorded, have the packet stream processing unit process the received packet.

With the configuration as described above, according to the information recording control device according to the present embodiment, it is determined whether or not the received packet is a known packet stream to be recorded, and the received packet is seconded based on the determination result. Have one of the session processing unit and the packet stream processing unit process. When the packet stream corresponds to the call control information, the first session processing unit determines that the recording target is to be recorded, inputs the information related to the packet stream to the packet stream processing unit, and holds the packet stream. Based on the information about the packet stream to be recorded, which is managed by the packet stream processing unit, it is determined that the received packet constitutes a known packet stream to be recorded, and the packet stream processing unit is made to process it. The packet stream processing unit collects received packets as main information packets and inputs them to the main information recording unit. The main information recording unit generates the main information and causes the recording device to record it.

On the other hand, when the packet stream does not correspond to the call control information, the packet stream processing unit does not yet have the information about the packet stream, and therefore determines that the received packet does not constitute a known packet stream to be recorded. Then, let the second session processing unit process it. The second session processing unit determines whether or not the received packet constitutes an unknown recording target packet stream, and as a result, if it is determined that the received packet is an unknown recording target packet stream, the second session processing unit outputs information about the packet stream. Input to the packet stream processing unit and hold it. Therefore, it is determined that the subsequent received packets constitute a known packet stream to be recorded, and the packet stream processing unit is made to process them. The packet stream processing unit collects received packets as main information packets and inputs them to the main information recording unit. The main information recording unit generates the main information and causes the recording device to record it. As a result, the main information can be comprehensively recorded regardless of whether or not the packet stream corresponds to the call control information.

(First Embodiment)
Hereinafter, a first embodiment in which the information recording control device of the present invention is applied to a call recording system that records a voice call in a VoIP environment will be described. In the following description, a call recording device will be illustrated as an information recording control device.

<System>
FIG. 1 is a schematic view showing an overall configuration of a call recording system including a call recording device according to the first embodiment. As shown in FIG. 1, in the call recording system 1, the gateway (GW) 4 is communicably connected to the optical communication network 2 via the optical network unit (ONU) 3.

The gateway 4 is a network device that bridges a fixed line and a subscriber side network in a communication service using a fixed line such as an optical communication network 2.

The gateway 4 is an example of a NAT device and has a built-in NAT function.

A switching hub 21 (21') provided in the first extension network 5 and the second extension network 6 is connected to the gateway 4. The first extension network 5 and the second extension network 6 will be described later.

The switching hubs 21 and 21'include mirror ports 21a and 21a'. The switching hubs 21 and 21'at the same time transmit data transmitted and received by the ports to which the first extension network 5 and the second extension network 6 are connected from the mirror ports 21a and 21a'.

A call recording device 7 is communicably connected to the mirror ports 21a and 21a', and is configured to receive IP packets flowing through the switching hubs 21 and 21'.

A switching hub 8 is communicably connected to the call recording device 7. A voice server 9 and PCs 10a to 10c are communicably connected to the switching hub 8.

The voice server 9 receives the voice file generated by the call recording device 7 and stores it in the storage device. The voice server 9 manages the voice file stored in the storage device in association with the recording-related information. The recording-related information is, for example, the telephone number of the calling side, the telephone number of the called side, the recording start time, and the length of the recording (call) time.

The PCs 10a to 10c are configured to request the voice server 9 to search for a voice file and play back a voice file transmitted from the voice server 9.

In the present embodiment, the voice server 9 is installed in the same local area network as the call recording device 7, but is installed outside the local area network, for example, via an Internet network (not shown). You may.

An external telephone terminal 11 is communicably connected to the optical communication network 2.

<Extension network>
FIG. 2 is a schematic view showing a first extension network and a second extension network in the call recording system shown in FIG. First, the first extension network 5 will be described. As shown in FIG. 2, the first extension network 5 includes a switching hub 21. The switching hub 21 is communicably connected to the port of the gateway 4 of FIG.

The SIP server 22 and the in-station telephone terminals 23a to 23c of FIG. 1 are connected to the plurality of ports of the switching hub 21.

The SIP server 22 is a server that manages and controls an IP telephone service using SIP, manages user identification information such as a telephone number and corresponding data of the location on the network such as an IP address, and and a caller. Mediates call information between call destinations.

The in-station telephone terminals 23a to 23c are SIP-compatible devices and function as SIP UAs (User Agents). The in-station telephone terminals 23a to 23c are IP telephones, IP telephone adapters, personal computers (PCs) and smartphones on which SIP-compatible applications operate, but are not particularly limited.

When the in-station telephone terminals 23a to 23c are connected to the switching hub 21, their telephone numbers and IP addresses are notified to the SIP server 22. Further, the in-station telephone terminals 23a to 23c notify the SIP server 22 of the telephone number of the telephone terminal on the called side at the time of making a call. The SIP server 22 transfers the outgoing call notification to the IP address of the calling side's telephone terminal corresponding to the notified telephone number, and when the communication start is permitted (the operation of "picking up the handset"), the SIP server 22 arrives at the calling side. The caller connects directly and starts a call.

As shown in FIG. 2, the second extension network 6 includes a switching hub 21', a SIP server 22', and an intra-station telephone terminal 23a' to 23c', and the switching hub 21 and the SIP server 22 of the first extension network 5 described above. , Has the same configuration and function as the in-station telephone terminals 23a to 23c.

The difference between the first extension network 5 and the second extension network 6 is that the call recording device 7 can use the call control information in the SIP message transmitted by the SIP server 22 of the first extension network 5, but the second extension. The point is that the call recording device 7 cannot use the call control information in the SIP message sent by the SIP server 22'of the network 6.

There are various reasons for this, but in this embodiment, the service provided by the SIP server 22'operated by the second extension network 6 has been changed, and the software version running on the SIP server 22' has been upgraded. It is assumed that the call recording device 7 cannot acquire call control information from call control packets transmitted / received in the SIP session of the SIP server 22'.

<Server>
FIG. 3 is a block diagram showing the hardware of the server constituting the call recording device according to the first embodiment.

As shown in FIG. 3, the server 100 includes a central control unit 101, a network control unit 102, a storage device 103, and a memory 104.

The central control unit 101 is a processor that controls each unit of the server 100. Specifically, the central control unit 101 includes a CPU and the like, and uses the memory 104 to perform various control operations according to the OS and various processing programs.

The network control unit 102 carries out data communication and the like via the network.

The storage device 103 is composed of, for example, a hard disk drive, a non-volatile memory, or the like, and stores various programs and data necessary for the operation of the central control unit 101.

The memory 104 is configured by, for example, a DRAM or the like, and functions as a buffer memory for temporarily storing data or the like processed by the central control unit 101 or the like, or a working memory of the central control unit 101 or the like.

In the server 100 described above, the call recording device 7 can be realized by appropriately preparing programs and data according to the purpose and causing the central control unit 101 to execute the programs and data. The SIP servers 22 and 22'of FIG. 2 and the voice server 9 of FIG. 1 can also be realized by the server 100.

<Call recording device>
FIG. 4 is a functional block diagram showing a call recording device according to the first embodiment. As shown in FIG. 4, in the call recording device 7, the packet capture unit 41 realized by the central control unit 101 (see FIG. 3) controls the network control unit 102 (see FIG. 3), and the switching hub 21 of FIG. 1 , 21'to receive IP packets from the mirror ports 21a, 21a'. The packet capture unit 41 outputs the received IP packet (hereinafter referred to as a received packet) to the received packet distribution unit 42 realized by the central control unit 101.

The reception packet distribution unit 42 outputs the input received packet to any one of the SIP session unit 43, the RTP session unit 44, and the RTP stream unit 45 realized by the central control unit 101 according to the determination flow described later.

The SIP session unit 43 is an example of the first session processing unit of the present invention, identifies a call control packet including a SIP message from a received packet, analyzes the call control packet, and makes a call included in the SDP of the SIP message. The calling side address indicating the address of the side node and the calling side address indicating the IP address of the calling side node are acquired.

The RTP session unit 44 is an example of the second session processing unit of the present invention, identifies an RTP packet from a received packet, analyzes the RTP packet, and acquires a destination address, a source address, and the like included in the IP header. To do.

The RTP stream unit 45 is an example of the packet stream processing unit of the present invention. The RTP stream unit 45 receives the calling side address and the calling side address acquired by the SIP session unit 43, and registers them in the RTP stream map as information regarding the packet stream to be recorded. The RTP stream map is stored by the central control unit 101 in the storage device 103 (see FIG. 3) or the memory 104 (see FIG. 3), which is an example of the storage device. The RTP stream map is configured to register a source address (srcaddr) and a destination address (dsdaddr) of a packet stream to be recorded as a pair. The RTP stream unit 45 uses the calling side address acquired by the SIP session unit 43 as the destination address and the incoming calling side address as the source address, and the upstream packet stream and the calling side address acquired by the SIP session unit 43. Is registered in the RTP stream map as the packet stream to be recorded, and the downlink packet stream is registered in the RTP stream map as the source address and the called party address as the destination address.

Further, the RTP stream unit 45 receives the destination address and the source address acquired by the RTP session unit 44, and registers them in the RTP stream map as information regarding the packet stream to be recorded.

Further, the RTP stream unit 45 detects and collects RTP packets constituting a packet stream in which the destination address and the transmission address registered in the RTP stream map match from the IP packet input from the received packet distribution unit 42. Voice data is taken out from the RTP packet and sequentially output to the voice queue 46.

The voice queue 46 is, for example, a queue prepared in the memory 104 by the central control unit 101, and is configured to sequentially output the voice data input earlier.

The call recording device 7 includes a recording channel management unit 47 that manages at least one voice channel. The recording channel management unit 47 is realized by the central control unit 101, and for example, a storage device 103, which is an example of a storage device, provides a usable recording area, that is, a recording channel.

The audio file generation unit 48 is realized by the central control unit 101, sequentially extracts audio data from the audio queue 46, converts it into audio data, and outputs it to a recording channel prepared by the recording channel management unit 47 in the form of an audio file. Remember. Further, the audio file generation unit 48 obtains and stores the recording start time and the recording (call) time from the time when the generation of the audio file starts and the time when the generation ends.

The audio file transmission unit 49 is realized by the central control unit 101, and transmits the audio file generated by the audio file generation unit 48 to the audio server 9. The voice file transmission unit 49 transmits recording-related information including the recording start time and the recording (call) time to the voice server 9 together with the voice file.

As described above, the RTP stream unit 45 causes the audio file generation unit 48 to generate audio data of both upstream and downlink packet streams from the collected RTP packets, and the audio file transmission unit 49 generates the audio file in FIG. 1. It is stored in the voice server 9 of. Therefore, the audio file generation unit 48 and the audio file transmission unit 49 correspond to the main information recording unit of the present embodiment.

<Distribution of received packets>
Next, the processing in the received packet distribution unit 42 (see FIG. 4) will be described. FIG. 5 is a flowchart showing processing in the received packet distribution unit in the call recording device according to the first embodiment. As shown in FIG. 5, the received packet distribution unit 42 first determines whether or not a packet has been received (S1). If the packet is received, it is determined whether or not the received packet is a call control packet (S2). This determination is performed, for example, by whether or not a character string peculiar to the SIP call control packet such as "SIP / 2.0" is included at the beginning or end of the first line. However, the determination is not limited to this example. If the determination is Yes in step S2, the received packet distribution unit 42 outputs the received packet to the SIP session unit 43 (S3).

If the determination is No in step S2, the received packet distribution unit 42 determines whether or not the packet is an RTP packet constituting a known packet stream to be recorded (S4). In this determination, for example, information about the packet stream to be recorded registered in the RTP stream map held by the RTP stream unit 45 using the source address (srcaddr) and the destination address (dsdaddr) included in the RTP packet as keys is used. The search is performed based on whether or not the information already exists. If the determination is Yes in step S4, the received packet distribution unit 42 outputs the received packet to the RTP stream unit 45 (S5).

If the determination in step S4 is No, the received packet distribution unit 42 outputs the received packet to the RTP session unit 44 (S6). After outputting the received packet to any of the SIP session unit 43, the RTP session unit 44, and the RTP stream unit 45 in steps S3, S5, and S6, the process returns to step S1 and waits for the next packet.

<RTP session processing>
Next, processing (hereinafter, also referred to as RTP session processing) in the RTP session unit 44 (see FIG. 4) will be described. FIG. 6 is a flowchart showing processing in the RTP session unit in the call recording device according to the first embodiment. As shown in FIG. 6, the RTP session unit 44 first determines whether or not the received packet seems to be an RTP packet (S11). This determination is performed based on conditions such as whether or not the received packet includes the version number included in the RTP header and whether or not the received packet has a data size equal to or larger than the header size.

If the determination in step S11 is No, the process ends. If the determination in step S11 is Yes, the RTP session unit 44 determines whether or not an RTP session exists (S12). One RTP session is created for a group of received packets having the same source address and destination address. Normally, since there are upstream and downstream packet streams for one call, two RTP sessions are generated in one call. When the determination in step S12 is No, that is, when the RTP session unit 44 receives the first received packet among the plurality of received packets having the same pair of source address and destination address, the RTP session Is newly created (S13). At this time, the RTP session unit 44 generates information about the RTP session and stores it in the storage device 103 or the memory 104 (see FIG. 2), which is an example of the storage unit. Information about the RTP session includes, for example, the source address and destination address, as well as the generation time, the final sequence number of the received packet (seq #), the synchronous source (SSRC) identifier of the first received packet, and the continuous counter. Includes a value. The initial value of the continuous counter value is zero. After completing the process of step S13, the process ends.

On the other hand, when the determination in step S12 is Yes, the RTP session unit 44 determines whether or not the SSRC identifier included in the received packet is the same as the SSRC identifier included in the information related to the RTP session (S14). If the determination in step S14 is No, the final sequence number and the continuous counter value are reset (S15), and the process ends.

If the determination in step S14 is Yes, it is determined whether or not the sequence number of the received packet is continuous with the final sequence number (S16). If the determination in step S16 is No, the process ends. On the other hand, if the determination in step S16 is Yes, the final sequence number is updated and the continuous counter value is incremented by 1 (S17). Next, the RTP session unit 44 determines whether or not the continuous counter value exceeds the threshold value (for example, 3) (S18).

If the determination in step S18 is Yes, the packet stream corresponding to the current RTP session is determined to be the packet stream to be recorded, information about the packet stream is output to the RTP stream unit 45, and call recording is started ( S19). After that, the RTP session unit 44 releases the current RTP session (S20) and ends the process.

On the other hand, if the determination in step S18 is No, it is determined whether or not a certain time has elapsed since the first received packet was received and a new RTP session was created (S21). If the determination in step S21 is No, the process ends. On the other hand, when the determination in step S21 is Yes, the RTP packet below the threshold value is received but does not continue, so the RTP session is released (S20) and the process is terminated.

<Call recording procedure>
Here, an example is a case where a call is recorded between the in-station telephone terminal 23a of the first extension network 5 (see FIG. 2) and the external telephone terminal 11 which is one of the nodes connected to the optical communication network 2. It will be explained by listing in.

FIG. 7 is a sequence diagram showing a call recording procedure in the call recording device according to the first embodiment. As shown in FIG. 7, first, when the in-station telephone terminal 23a performs the operation of calling the external telephone terminal 11 (S101), the INVITE request for calling the external telephone terminal 11 is sent to the SIP server 22 of the first extension network 5. It is transmitted to (see FIG. 2) (S102). In FIG. 7, “INVITE W / A” indicates that the call control packet constituting the INVITE request includes the IP address (A) of the station telephone terminal 23a. The SIP server 22 transfers the INVITE request received from the in-station telephone terminal 23a to the external telephone terminal 11 (S103). In the call recording device 7, the packet capture unit 41 takes in the call control packet constituting the INVITE request from the mirror ports 21a and 21a'of the switching hubs 21 and 21'in FIG. 1, and outputs the call control packet to the reception packet distribution unit 42. The reception packet distribution unit 42 outputs the call control packet to the SIP session unit 43. The SIP session unit 43 analyzes the call control packet of the INVITE request and acquires the IP address (A) of the in-station telephone terminal 23a as the calling side address (S104).

When the handset is picked up as a response process in response to the INVITE request on the external telephone terminal 11 (S105), a 200 OK response (denoted as 200 / INVITE in FIG. 7) is transmitted to the SIP server 22 (S106). In FIG. 7, “200 / INVITE W / B” indicates that the call control packet constituting the 200 OK response includes the IP address (B) of the external telephone terminal 11.

In the call recording device 7, the packet capture unit 41 takes in the call control packet constituting the 200 OK response from the mirror ports 21a and 21a'of the switching hubs 21 and 21'in FIG. 1 and outputs the call control packet to the received packet distribution unit 42. The reception packet distribution unit 42 outputs the call control packet to the SIP session unit 43. The SIP session unit 43 analyzes the call control packet of the 200 OK response and acquires the IP address (B) of the external telephone terminal 11 as the incoming call side address (S107). The SIP server 22 transfers the 200 OK response received from the external telephone terminal 11 to the in-station telephone terminal 23a (S108).

A call is started between the in-station telephone terminal 23a and the external telephone terminal 11 using the IP addresses (A) and (B) of the other party exchanged with each other (S109). That is, the in-station telephone terminal 23a converts the voice of the speaker into voice data, further converts the voice data into an RTP packet in which the IP address (B) of the external telephone terminal 11 is set as the destination address, and sends the voice data ( S110). This is called a downlink packet stream. In FIG. 7, “A → B” indicates that the source address of the RTP packet is the IP address (A) and the destination IP address (B). On the other hand, the external telephone terminal 11 converts the voice of the speaker into voice data, further converts the voice data into an RTP packet in which the IP address (A) of the in-station telephone terminal 23a is set as the destination address, and sends the voice data ( S111). This is called an upstream packet stream. In FIG. 4, “B → A” indicates that the source address of the RTP packet is the IP address (B) and the destination IP address (A).

The call recording device 7 captures both upstream and downstream packet streams and records the call. That is, in the call recording device 7, first, the packet capture unit 41 outputs the received packet received from the mirror ports 21a, 21a'of the switching hubs 21 and 21'in FIG. 1 to the received packet distribution unit 42.

The reception packet distribution unit 42 determines that the received packet is an RTP packet constituting a known packet stream to be recorded, and outputs the received packet to the RTP stream unit 45.

From the received packet, the RTP stream unit 45 acquires an RTP packet whose destination addresses are the IP address (A) and the IP address (B), which constitute both the upstream and downstream packet streams detected by the SIP session unit 43. Then (S112, S113), voice data is taken out from the acquired RTP packet and sequentially output to the voice queue 46.

The audio file generation unit 48 sequentially converts audio data from the audio queue 46 into audio data, and outputs the audio data in the form of an audio file to the recording channel prepared by the recording channel management unit 47. The output audio file is transmitted by the audio file transmission unit 49 to the audio server 9.

Next, for example, in the external telephone terminal 11, when the handset is lowered (S114), a BYE request notifying the end of the call is transmitted to the SIP server 22 (S115). The SIP server 22 transfers the BYE request received from the external telephone terminal 11 to the in-station telephone terminal 23a (S116). Upon receiving the BYE request, the in-station telephone terminal 23a transmits a 200 OK response (denoted as 200 / BYE in FIG. 7) (S117). In the call recording device 7, the packet capture unit 41 takes in the call control packet constituting the 200 OK response from the mirror ports 21a and 21a'of the switching hubs 21 and 21'in FIG. 1, and outputs the call control packet to the reception packet distribution unit 42. The reception packet distribution unit 42 outputs a call control packet to the SIP session unit 43. The SIP server 22 transfers the 200 OK response received from the in-station telephone terminal 23a to the external telephone terminal 11 (S118). The SIP session unit 43 analyzes the call control packet of the 200 OK response, determines the end of the call, and ends the recording (S119). As a result, the recording can be terminated when the call between the in-station telephone terminal 23a and the external telephone terminal 11 is terminated (S120).

Next, the case of recording a call between the in-station telephone terminal 23a'of the second extension network 6 (see FIG. 2) and the external telephone terminal 11 which is one of the nodes connected to the optical communication network 2. explain. FIG. 8 is a sequence diagram showing a call recording procedure in the call recording device according to the first embodiment. In FIG. 8, the same procedure as in FIG. 7 is designated by the same reference numerals and the description thereof will be omitted.

The procedure from S101 to S109 to the start of the call is basically the same as described with reference to FIG. 7, under the control of the SIP server 22', the in-station telephone terminal 23a'and the external telephone terminal 11 Call control information is exchanged with and the call is started.

However, since the SIP server 22'uses an INVITE request and a 200 OK response call control packet that cannot be analyzed by the SIP session unit 43 of the call recording device 7, the IP address (A) of the in-station telephone terminal 23a'and the external telephone The IP address (B) of the terminal 11 cannot be acquired (S104'and S107'). As a result, the RTP stream unit 45 is in a state where the information regarding the packet stream to be recorded does not exist in the RTP stream map.

Therefore, when the received packet distribution unit 42 receives the received packets constituting the downlink packet stream (S110) and the uplink packet stream (S111) (see FIG. 8), it is known in step S4 described with reference to FIG. Since it is not a packet stream to be recorded, it is determined as No, the process proceeds to step S6, and the stream is output to the RTP session unit 44. As a result, the RTP session unit 44 executes the RTP session processing described with reference to FIG. 6 (S201 and S202 in FIG. 8).

FIG. 9 is an explanatory diagram showing RTP session processing for RTP packets constituting both the upstream and downstream packet streams shown in FIG. 8. As shown in FIG. 9, when the call recording device 7 receives the first RTP packet, the continuous counter value is incremented by 1 in step S17 in FIG. 6 to become 1. Similarly, when the 2nd to 4th RTP packets are received, the continuous counter value is incremented by 1 in step S17 in FIG. 6 and counted up to 2, 3, and 4. When the fourth RTP packet is received, the continuous counter value becomes 4, and the threshold value 3 is exceeded. Therefore, the determination in step S18 in FIG. 6 becomes Yes, and recording is started (S19 in FIG. 6).

As a result, the RTP session unit 44 can perform the processes of steps S11 to S18 shown in FIG. 6, output the source address and the destination address to the RTP stream unit 45, and start recording.

On the other hand, the SIP session unit 43 of the call recording device 7 cannot analyze the call control packet of the 200 OK response, determine the end of the call, and end the recording. Therefore, the RTP stream unit 45 estimates the end of the call, for example, on the condition that the next RTP packet is not input even after a certain period of time has passed since the latest RTP packet was input from the received packet distribution unit 42. Then, the recording is finished (S203).

<Effect>
According to the call recording device 7 according to the first embodiment described above, first, the SIP session unit 43 acquires call control information from the call control packet transmitted from the first extension network 5, and the recording target. Information about the packet stream is input to the RTP stream unit 45 and registered in the RTP stream map. The received packet distribution unit 42 refers to the RTP stream map, determines that the received packet constitutes a known packet stream to be recorded, and outputs the received packet to the RTP stream unit 45. The RTP stream unit 45 collects the received packet as an RTP packet and inputs it to the voice queue 46.

On the other hand, the packet stream transmitted from the second extension network 6 does not correspond to the call control information acquired by the SIP session unit 43. In this case, since the information about the packet stream is not yet registered in the RTP stream map of the RTP stream unit 45, the received packet distribution unit 42 determines that the received packet does not constitute a known packet stream to be recorded. , Output to the RTP session unit 44. The RTP session unit 44 determines whether or not the received packet constitutes an unknown recording target packet stream by the RTP session processing shown in FIG. 6, and as a result, determines that the received packet is an unknown recording target packet stream. In this case, information about the packet stream is input to the RTP stream unit 45 and registered in the RTP stream map. Therefore, the received packet distribution unit 42 determines that the subsequent received packets constitute a known packet stream to be recorded, and outputs the subsequent received packets to the RTP stream unit 45. The RTP stream unit 45 collects the received packet as an RTP packet and inputs it to the voice queue 46.

As a result, according to the call recording device 7 according to the first embodiment, it is possible to record the packet stream transmitted from either the first extension network 5 or the second extension network 6. It is possible to suppress the occurrence of omissions and comprehensively record calls.

In particular, according to the call recording device 7 according to the first embodiment, the SIP session unit 43 analyzes the call control packet transmitted by the SIP server 22 for the first extension network 5, and the call control information can be used. Therefore, it is possible to grasp the start and end timings of the call and record the call with high accuracy. In addition, additional information such as a call start time, end time, and arrival / departure number can be acquired from the call control packet. As described above, the call recording device 7 according to the first embodiment mainly records a call based on the call control information, and utilizes the call control information to perform high-precision and high-value-added call recording. When call control information is not available, autonomous recording can be performed to achieve comprehensive call recording.

In the first embodiment, since the version of the software running on the SIP server 22'of the second extension network 6 has been upgraded, the call recording device 7 sends and receives call control packets in the SIP session of the SIP server 22'. The case where the call control information cannot be obtained from is described as an example. However, the present invention is not limited to this, and the present invention can be applied to, for example, the following cases.

(A) Depending on the network configuration, the call control packet transmitted by the second extension network 6 does not flow to the network where the call recording device 7 is located, but only the RTP packet may flow. According to the call recording device 7 according to the first embodiment, even in such a case, the RTP session processing (see FIG. 6) analyzes the RTP packet, identifies the packet stream to be recorded, and records the call. It can be performed.

(B) There are cases where the second extension network 6 uses a protocol other than SIP (for example, H323 or a protocol unique to the SIP server vendor). In the case of a vendor-specific protocol, the specifications are often not disclosed, and there is a limit for the vendor of the call recording device 7 to clarify the protocol by reverse engineering or the like and follow it by changing the design of the call recording device 7 or adding functions. There is. According to the call recording device 7 according to the first embodiment, even in such a case, the RTP session processing (see FIG. 6) analyzes the RTP packet, identifies the packet stream to be recorded, and records the call. It can be performed.

(C) There is a case where the SIP control process in the second extension network 6 is complicated. Similar to the above, when the call control information is changed due to the addition of a service in the second extension network 6, the call control information is unique to the vendor, the specifications are often not disclosed, and the call is made. There is a limit to how the vendor of the recording device 7 can clarify the details of the SIP control process by reverse engineering or the like and follow up with the design change or function addition of the call recording device 7. The vendor of the call recording device 7 cannot control the upgrade or setting change of the software running on the SIP server 22'. According to the call recording device 7 according to the first embodiment, even in such a case, the RTP session processing (see FIG. 6) analyzes the RTP packet, identifies the packet stream to be recorded, and records the call. It can be performed.

(D) There are cases where a special call pattern is used in the second extension network 6 other than the vendor's own SIP control process or the version or setting of the software running on the SIP server 22'. For example, even if the SIP control process is within the standard specifications, a complicated operation mode may occur in the SIP server 22'. In such a case, there is a limit for the vendor of the call recording device 7 to elucidate a special call pattern by reverse engineering or the like and follow it by changing the design of the call recording device 7 or adding a function. The vendor of the call recording device 7 cannot control the change of the operation mode of the SIP server 22'. According to the call recording device 7 according to the first embodiment, even in such a case, the RTP session processing (see FIG. 6) analyzes the RTP packet, identifies the packet stream to be recorded, and records the call. It can be performed.

(E) There is also a system that transmits only RTP packets on the second extension network 6. According to the call recording device 7 according to the first embodiment, even in such a case, the RTP session processing (see FIG. 6) analyzes the RTP packet, identifies the packet stream to be recorded, and records the call. It can be performed.

(Second Embodiment)
Hereinafter, a second embodiment in which the information recording control device of the present invention is applied to a call recording system that records a voice call in a VoIP environment will be described. In the following description, a call recording device will be illustrated as an information recording control device. The second embodiment differs from the first embodiment in that the call recording device acquires call control information from the PBX.

A second embodiment will be described with reference to FIGS. 10 to 15. 10 to 15, the same configurations as those used in the first embodiment are designated by the same reference numerals as those used in FIGS. 1 to 9, and the description thereof may be omitted.

<System>
FIG. 10 is a schematic view showing the overall configuration of a call recording system including the call recording device according to the second embodiment. As shown in FIG. 10, the call recording system 1'provides a first extension network 5'and a second extension network 6'operated by the PBX, and a call recording device 7'providing a function of linking with the PBX. There is. The first extension network 5', the second extension network 6', and the call recording device 7'will be described later.

<Extension network>
FIG. 11 is a schematic view showing a first extension network and a second extension network in the call recording system shown in FIG. First, the first extension network 5'will be described. As shown in FIG. 11, the PBX 51 is provided. The PBX51 is a server that manages and controls the IP telephone service, manages user identification information such as a telephone number and correspondence data of the location on the network such as an IP address, and makes a call between the caller and the callee. Mediate call information. In the present embodiment, PBXs 51 and 51'use SIP for call control, but are not particularly limited. The in-station telephone terminals 23a to 23c are configured to enable a call on an outside line and an extension line under the control of the PBX 51.

Further, the PBX 51 is configured to be able to communicate with each other via a network with the PBX cooperation unit 61 of the call recording device 7'described later.

As shown in FIG. 11, the second extension network 6'includes a switching hub 21', PBX51', and in-station telephone terminals 23a' to 23c', and the switching hub 21, PBX51, in the station of the first extension network 5'described above. It has the same configuration and function as the telephone terminals 23a to 23c.

The difference between the first extension network 5'and the second extension network 6'is that the PBX51 of the first extension network 5'can cooperate with the call recording device 7', but the PBX51'of the second extension network 6' , It is impossible to cooperate with the call recording device 7'.

There are various reasons for this, but in the present embodiment, the service provided by the PBX51 operated on the second extension network 6'has changed, and the version of the software running on the PBX51' has been upgraded, so that the call recording device It is assumed that 7'cannot cooperate with PBX51'.

The PBXs 51 and 51'can be realized by appropriately preparing programs and data according to the purpose in the server 100 described with reference to FIG. 3 and causing the central control unit 101 to execute the programs and data.

<Call recording device>
FIG. 12 is a functional block diagram showing a call recording device according to the second embodiment. As shown in FIG. 12, the call recording device 7'is different in that it includes a PBX cooperation unit 61 instead of the SIP session unit 43 shown in FIG. 4 as an example of the first session processing unit of the present invention. ..

The PBX cooperation unit 61 is realized by the central control unit 101 (see FIG. 3), controls the network control unit 102, and is configured to be able to communicate with the PBXs 51 and 51'via the network. The PBX cooperation unit 61 associates the extension numbers of the in-station telephone terminals 23a to 23c and 23a'to 23c'and the external line numbers of the external telephone terminals 11 with the IP address of each terminal, and serves as a terminal setting file as an example of the storage unit. It is held in a certain storage device 103 or memory 104.

Further, the PBX cooperation unit 61 receives the call control information from the PBXs 51 and 51', and outputs, for example, the calling side address and the called side address, which are information about the packet stream included in the call control information, to the RTP stream unit 45. Then, register it in the RTP stream map.

<Distribution of received packets>
Next, the processing in the received packet distribution unit 42'will be described. FIG. 13 is a flowchart showing processing in the received packet distribution unit in the call recording device according to the second embodiment. As shown in FIG. 13, the received packet distribution unit 42'first determines whether or not a packet has been received (S21). When the packet is received, the received packet distribution unit 42'determines whether or not the packet is an RTP packet constituting a known packet stream to be recorded (S22). In this determination, for example, information about the packet stream to be recorded registered in the RTP stream map held by the RTP stream unit 45 using the source address (srcaddr) and the destination address (dsdaddr) included in the RTP packet as keys is used. The search is performed based on whether or not the information already exists. If the determination is Yes in step S22, the received packet distribution unit 42'outputs the received packet to the RTP stream unit 45 (S23).

If the determination in step S22 is No, the received packet distribution unit 42'outputs the received packet to the RTP session unit 44 (S24). After outputting the received packet to either the RTP session unit 44 or the RTP stream unit 45 in steps S23 and S24, the process returns to step S21 and waits for the next packet.

The call recording device 7'is the same as the call recording device 7 described with reference to FIG. 4 in other configurations.

<Call recording procedure>
Here, an example is a case where a call is recorded between the in-station telephone terminal 23a of the first extension network 5 (see FIG. 10) and the external telephone terminal 11 which is one of the nodes connected to the optical communication network 2. It will be explained by listing in.

FIG. 14 is a sequence diagram showing a call recording procedure in the call recording device according to the second embodiment. As shown in FIG. 14, first, when the in-station telephone terminal 23a performs the operation of calling the external telephone terminal 11 (S101), the INVITE request for calling the external telephone terminal 11 is the PBX51 of the first extension network 5'(S101). (See FIG. 11) (S102). The PBX 51 forwards the INVITE request received from the in-station telephone terminal 23a to the external telephone terminal 11 (S103).

When the handset is picked up as a response process in response to the INVITE request on the external telephone terminal 11 (S105), a 200 OK response (denoted as 200 / INVITE in FIG. 14) is transmitted to the PBX 51 (S106).

When the PBX 51 receives the 200 OK response, the extension number of the in-station telephone terminal 23a (for example, 2001) is used as the telephone number of the calling terminal, and the external number of the external telephone terminal 11 is used as the telephone number of the calling terminal. Call control information including (for example, 03-1234-5678) is transmitted to the PBX cooperation unit 61 of the call recording device 7'. Further, the PBX 51 transfers the 200 OK response received from the external telephone terminal 11 to the in-station telephone terminal 23a (S108).

In the call recording device 7', the PBX cooperation unit 61 refers to the terminal setting file and acquires the IP address (A) corresponding to the extension number of the in-station telephone terminal 23a included in the call control information as the calling side address. Further, the PBX cooperation unit 61 refers to the terminal setting file and acquires the IP address (B) corresponding to the outside line number of the external telephone terminal 11 included in the call control information as the incoming call side address.

A call is started between the in-station telephone terminal 23a and the external telephone terminal 11 using the IP addresses (A) and (B) of the other party exchanged with each other (S109). That is, the in-station telephone terminal 23a converts the voice of the speaker into voice data, further converts the voice data into an RTP packet in which the IP address (B) of the external telephone terminal 11 is set as the destination address, and sends the voice data ( S110). This is called a downlink packet stream. On the other hand, the external telephone terminal 11 converts the voice of the speaker into voice data, further converts the voice data into an RTP packet in which the IP address (A) of the in-station telephone terminal 23a is set as the destination address, and sends the voice data ( S111). This is called an upstream packet stream.

The call recording device 7'captures both upstream and downstream packet streams and records the call. That is, in the call recording device 7', the packet capture unit 41 first outputs the received packets received from the mirror ports 21a and 21a'of the switching hubs 21 and 21' in FIG. 10 to the received packet distribution unit 42'.

The reception packet distribution unit 42'determines that the received packet is an RTP packet constituting a known packet stream to be recorded, and outputs the received packet to the RTP stream unit 45.

From the received packet, the RTP stream unit 45 acquires the RTP packet whose destination addresses are the IP address (A) and the IP address (B), which constitute both the upstream and downstream packet streams acquired by the PBX cooperation unit 61. (S112, S113), voice data is taken out from the acquired RTP packet and sequentially output to the voice queue 46.

The audio file generation unit 48 sequentially converts audio data from the audio queue 46 into audio data, and outputs the audio data in the form of an audio file to the recording channel prepared by the recording channel management unit 47. The output audio file is transmitted by the audio file transmission unit 49 to the audio server 9.

Next, for example, in the external telephone terminal 11, when the handset is lowered (S114), a BYE request notifying the end of the call is transmitted to the PBX 51 (S115). The PBX 51 forwards the BYE request received from the external telephone terminal 11 to the in-station telephone terminal 23a (S116). Upon receiving the BYE request, the in-station telephone terminal 23a transmits a 200 OK response (denoted as 200 / BYE in FIG. 14) (S117). The PBX 51 transfers the 200 OK response received from the in-station telephone terminal 23a to the external telephone terminal 11 (S118). Further, the PBX 51 transmits call control information indicating that the call between the extension number 2001 and the outside line number 03-1234-5678 has been completed to the PBX cooperation unit 61 (S302). The PBX cooperation unit 61 determines the end of the call and ends the recording (S303). As a result, the recording can be terminated when the call between the in-station telephone terminal 23a and the external telephone terminal 11 is terminated (S120).

Next, when recording a call between the in-station telephone terminal 23a'of the second extension network 6'(see FIG. 11) and the external telephone terminal 11 which is one of the nodes connected to the optical communication network 2. Will be described. FIG. 15 is a sequence diagram showing a call recording procedure in the call recording device according to the first embodiment. In FIG. 15, the same procedure as in FIG. 14 is designated by the same reference numerals and the description thereof will be omitted.

The procedure up to the start of the call shown in S101 to S103, S105, S106, S108, and S109 is basically the same as described with reference to FIG. 15, under the control of the PBX51', the station telephone terminal 23a. 'And the external telephone terminal 11 exchanges call control information, and a call is started.

However, since the PBX 51'uses call control information that cannot be analyzed by the PBX cooperation unit 61 of the call recording device 7', the IP address (A) of the in-station telephone terminal 23a'and the IP address (B) of the external telephone terminal 11 are used. Cannot be obtained. As a result, the RTP stream unit 45 is in a state where the information regarding the packet stream to be recorded does not exist in the RTP stream map.

Therefore, when the received packet distribution unit 42'receives the received packets constituting the downlink packet stream (S110) and the uplink packet stream (S111) (see FIG. 15), it is known in step S22 described with reference to FIG. Since it is not the packet stream to be recorded, it is determined that No, the process proceeds to step S24, and the stream is output to the RTP session unit 44. As a result, the RTP session unit 44 executes the RTP session processing described with reference to FIG. 6 (S311 and S312 in FIG. 15).

As a result, the RTP session unit 44 can perform the processes of steps S11 to S18 shown in FIG. 6, output the source address and the destination address to the RTP stream unit 45, and start recording.

On the other hand, the PBX cooperation unit 61 of the call recording device 7'is unable to receive the call control information 302 for the end of the call, determine the end of the call, and end the recording. Therefore, for example, the RTP stream unit 45 terminates the call on the condition that the next RTP packet is not input even after a certain period of time has passed since the latest RTP packet was input from the received packet distribution unit 42'. Estimate and end recording (S313).

<Effect>
According to the call recording device 7'according to the second embodiment described above, first, the PBX cooperation unit 61 records a packet stream to be recorded based on the call control information transmitted by the PBX 51 of the first extension network 5'. Information about the above is input to the RTP stream unit 45 and registered in the RTP stream map. The received packet distribution unit 42'refers to the RTP stream map, determines that the received packet constitutes a known packet stream to be recorded, and outputs the received packet to the RTP stream unit 45. The RTP stream unit 45 collects the received packet as an RTP packet and inputs it to the voice queue 46.

On the other hand, the packet stream transmitted from the second extension network 6'does not correspond to the call control information received by the PBX cooperation unit 61. In this case, since the information about the packet stream is not yet registered in the RTP stream map of the RTP stream unit 45, the received packet distribution unit 42'determines that the received packet does not constitute a known packet stream to be recorded. Then, it is output to the RTP session unit 44. The RTP session unit 44 determines whether or not the received packet constitutes an unknown recording target packet stream by the RTP session processing shown in FIG. 6, and as a result, determines that the received packet is an unknown recording target packet stream. In this case, information about the packet stream is input to the RTP stream unit 45 and registered in the RTP stream map. Therefore, the received packet distribution unit 42'determines that the subsequent received packets constitute a known packet stream to be recorded, and outputs the subsequent received packets to the RTP stream unit 45. The RTP stream unit 45 collects the received packet as an RTP packet and inputs it to the voice queue 46.

As a result, according to the call recording device 7'according to the second embodiment, it is possible to record the packet stream transmitted from either the first extension network 5'or the second extension network 6'. Therefore, it is possible to suppress the occurrence of omissions and comprehensively record calls.

In particular, according to the call recording device 7'according to the second embodiment, the call control packet transmitted by the PBX 51 is received by the PBX cooperation unit 61 for the first extension network 5', and the call control information can be used. Therefore, it is possible to grasp the start and end timings of the call and record the call with high accuracy. In addition, additional information such as a call start time, end time, and departure / arrival number can be acquired from the PBX. As described above, the call recording device 7'according to the second embodiment mainly performs call recording based on the call control information, and utilizes the call control information to perform high-precision and high-value-added call recording. , When call control information is not available, autonomous recording can be performed to achieve comprehensive call recording.

In the second embodiment described above, the case where the call recording device 7'is provided with the PBX cooperation unit 61 as the first session processing unit of the present invention has been described as an example, but the function of the PBX cooperation unit is executed. A PBX cooperation device composed of a server 100 (see FIG. 3) provided with a central control unit 101 is provided separately from the call recording device 7', and call control information transmitted from the PBX 51, 51'is used as the PBX cooperation device. The PBX cooperation device may be configured to execute the same processing as the above-mentioned PBX cooperation unit 61 and transmit the processing result to the call recording device 7'. In this case, in the call recording device 7', among the processes of the PBX cooperation unit 61 described above, the calling side address and the called side address, which are information about the packet stream included in the call control information from the PBXs 51 and 51', are transferred to the RTP stream. The processing unit that performs the processing of outputting to the unit 45 and registering it in the RTP stream map and interlocking with the PBX interlocking device exists as the first session processing unit of the present invention.

Although the present invention has been described in detail above, it is clear to those skilled in the art that the present invention is not limited to the embodiments described in the present specification. The present invention can be implemented as modifications and modifications without departing from the spirit and scope of the present invention as defined by the claims. Therefore, the description of the present specification is for the purpose of exemplification and does not have any limiting meaning to the present invention.

In the above-described embodiment, the received packet distribution units 42 and 42'(see FIGS. 4 and 12) are provided, and the RTP packet constituting the packet stream of the known recording target is sent to the RTP session unit 44 of the unknown recording target. The remaining received packets including the RTP packets constituting the packet stream are selectively output to the RTP session unit 44, and each unit is made to process the received packet. However, for example, the received packet is copied and input to both the RTP session section 44 and the RTP stream section 45, and the RTP session section 44 refers to the RTP stream map of the RTP stream section 45 and is known as a packet stream to be recorded. RTP packets constituting the above may be identified and discarded, RTP session processing (see FIG. 6) may not be performed, and RTP session processing (see FIG. 6) may be performed on other received packets. Similarly, the RTP stream unit 45 refers to the RTP stream map, identifies and discards received packets other than the RTP packets constituting the known recording target packet stream, and constitutes the known recording target packet stream. You may want to process the packet.

In the above-described embodiment, when a packet stream that does not correspond to the call control information is to be recorded, additional information cannot be acquired from the SIP servers 22 and 22'(see FIG. 2) and PBX 51 and 51'(see FIG. 11). In the call recording devices 7 and 7'or the voice server 9 (see FIGS. 1 and 10), the central control unit 101 of the server 100 constituting these is connected to the in-station telephone terminals 23a to 23c and 23a'to 23c'(FIG. 2, FIG. Storage that is a storage unit in the form of a terminal setting file by associating the IP address of each node of the external telephone terminal 11 (see FIGS. 1 and 10) with the telephone number (external line number or extension number). It is stored in the device 103 or the memory 104. When the RTP session unit 44 determines that an unknown packet stream is to be recorded, the terminal setting file is referred to using the source address and destination address of the RTP packet that composes the unknown packet stream, and the source address and destination address are used. The telephone numbers corresponding to the matching IP addresses can be specified as the telephone numbers of the calling side and the called side. The telephone numbers of the calling side and the called side identified in this way can be associated with the voice file as recording-related information and managed by the voice server 9.

Further, for example, in the call recording systems 1 and 1'according to the above-described embodiment, an illegal call such as using the security holes of the SIP servers 22 and 22'or the PBX 51 and 51'may be performed. In the call recording devices 7 and 7'according to the present embodiment, even in the packet stream of such an illegal call, the RTP session unit 44 autonomously detects and records the call as a recording target without using the call control information. It is possible to leave evidence that an unauthorized call has been made.

Further, in the call recording systems 1 and 1'according to the above-described embodiment, a call that remains connected may occur due to a malfunction of the SIP servers 22, 22'or PBX 51, 51'. In the call recording devices 7 and 7'according to the present embodiment, the RTP session constituting the packet stream transmitted and received by the continuously connected call is autonomously recorded by the RTP session unit 44 without using the call control information. Since it can be detected and a record can be recorded, it is possible to reduce wasteful resource use of the network and the server 100 (see FIG. 3) due to continuous calls.

Further, in the first embodiment, in the first extension network 5 and the second extension network 6, the SIP servers 22 and 22'are exchanging call control information (see FIGS. 1 and 2), and the first embodiment. In the second embodiment, examples in which the PBXs 51 and 51'are exchanging call control information in the first extension network 5'and the second extension network 6'(see FIGS. 10 and 11) have been described, respectively. The extension network of the above may be operated by the SIP server, and the other extension network may be operated by the PBX.

Further, in the above-described embodiment, the call recording devices 7 and 7'for recording a call on two extension networks have been described, but the present invention is also applied when recording a call on three or more extension networks. can do.

Further, in the above-described first embodiment, the central control unit 101 (see FIG. 3) uses the SIP session unit 43 (see FIG. 4) as the first session processing unit and the RTP session unit as the second session processing unit. Although the case where both the 44 and the RTP stream unit 45 which is the packet stream processing unit are realized has been described as an example, the SIP session unit 43, the RTP session unit 44, and the RTP stream unit 45 may be realized by different processors. Good.

Similarly, in the second embodiment described above, the PBX cooperation unit 61, the RTP session unit 44, and the RTP stream unit 45 may be realized by different processors.

In addition, the terms described herein and / or the terms necessary for understanding the present specification may be replaced with terms having the same or similar meanings.

The information, signals, etc. described herein may be represented using any of a variety of different techniques. For example, data, instructions, commands, information, signals, bits, symbols, chips, etc. that may be referred to throughout the above description are voltages, currents, electromagnetic waves, magnetic fields or magnetic particles, light fields or photons, or any of these. It may be represented by a combination of.

Each aspect / embodiment described in the present specification may be used alone, in combination, or may be switched and used according to the execution. Further, the notification of predetermined information (for example, the notification of "being X") is not limited to the explicit one, but is implicitly (for example, by not notifying the predetermined information). You may.

The order of the processing procedures, sequences, flowcharts, and the like of each aspect / embodiment described in the present specification may be changed as long as there is no contradiction. For example, the methods described herein present elements of various steps in an exemplary order, and are not limited to the particular order presented.

The present invention has an effect of enabling comprehensive information recording, and can be suitably used for a call recording device, a videophone recording device, and the like.

1, 1'Call recording system 2 Optical communication network 5, 5'First extension network 6, 6'Second extension network 7, 7'Call recording device 9 Voice server 10 Personal computer (PC)
11 External telephone terminals 21, 21'Gateways 22, 22' SIP servers 23a to 23c, 23a' to 23c' Internal telephone terminals 41 Packet capture unit 42, 42'Received packet distribution unit 43 SIP session unit 44 RTP session unit 45 RTP stream Part 46 Voice Queue 47 Recording Channel Management Unit 48 Voice File Generation Unit 49 Voice File Transmission Unit 61 PBX Linkage Unit

Claims (8)

The first session processing unit that determines the packet stream corresponding to the call control information included in the call control packet as the recording target, and
A second session processing unit that determines a packet stream that does not correspond to the call control information as a recording target,
A packet stream processing unit that collects the packet stream to be recorded determined by the first session processing unit and the second session processing unit, and a packet stream processing unit.
It is provided with a main information recording unit that generates main information from a plurality of main information packets that constitute the packet stream collected by the packet stream processing unit and causes a recording device to record the main information.
For each of the plurality of received packets flowing through the network, it is determined whether or not the received packet is a known main information packet constituting the packet stream to be recorded, and based on the determination result, the received packet is subjected to the second session processing unit. And an information recording control device characterized in that any one of the packet stream processing units is used for processing. The call control information, the information recording control apparatus according to claim 1, characterized in that the information about the node obtained from the call control packet. When the received packet is the call control packet, the received packet is output to the first session processing unit, and when the received packet is not a known packet stream to be recorded, the received packet is sent to the second session. The claim is characterized by further comprising a received packet distribution unit that outputs the received packet to the processing unit and outputs the received packet to the packet stream processing unit when the received packet is a known packet stream to be recorded. The information recording control device according to 2. The information recording control device according to any one of claims 1 to 3, wherein the call control information is information about a node acquired from a private telephone exchange. It is determined whether or not the received packet is a known call control packet based on information about the node acquired from the private telephone exchange, and if the received packet is not a known packet stream to be recorded, the received packet. Is output to the second session processing unit, and when the received packet is a known packet stream to be recorded, the received packet distribution unit that outputs the received packet to the packet stream processing unit is further provided. The information recording control device according to claim 4, which is characterized. The information recording control device according to any one of claims 1 to 5, wherein the main information packet is an RTP packet. The information recording control device according to any one of claims 1 to 6, wherein the main information is at least one selected from voice, image, video, and text. The first determination step of determining the packet stream corresponding to the call control information as the recording target, and
A second determination step of determining whether the received packet flowing through the network is a main information packet constituting the known packet stream to be recorded.
Based on the determination result, the third determination process for determining the received packet as the recording target for the packet stream that does not correspond to the call control information, and the plurality of main information packets constituting the packet stream are collected, and the plurality of main information packets are collected. The process of generating the main information from the main information packet and attaching it to any one of the packet stream processes to be recorded in the recording device, and
An information recording control method comprising.

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