JP2023117556A - Packet loss detection system and packet loss detection method - Google Patents

Abstract

To provide a packet loss detection system and a packet loss detection method capable of detecting packet loss at the start or end of a call when making the call through a network.SOLUTION: A packet loss detection system includes: a network communication unit that transmits and receives packets containing audio data via a network; an audio communication unit that converts audio signals into packets and converts the audio data into audio signals and outputs them; a transmitting/receiving unit that extracts the audio data from received packets that are packets received from the network to transmit it to the audio communication unit, and transmits the packets received from the audio communication unit to the network as transmission packets; and a determination unit that determines whether there is a loss in received packets. The transmitting/receiving unit extracts the number of transmitted packets and the number of received packets. The determination unit compares the number of transmitted packets with the number of received packets, and determines that there is a loss of received packets if the number of received packets is smaller than the number of transmitted packets.SELECTED DRAWING: Figure 1

Description

The present invention relates to a packet loss detection system and a packet loss detection method for detecting loss of packets transmitted and received via a network.

2. Description of the Related Art In recent years, with the spread of the Internet, an IP telephony service has come to be provided that enables voice calls by IP telephony using VoIP (Voice over Internet Protocol) technology on the Internet. IP telephones implement call control mainly using SIP (Session Initiation Protocol). In IP telephony, RTP packets, which are packets in which RTP (Real-time Transport Protocol) headers are added to voice data, are transmitted and received via the Internet.

However, IP telephones may lose RTP packets during communication due to failures in communication paths on the Internet or failures in communication equipment. In the IP telephony service, when packet loss (packet loss) occurs in which RTP packets are lost during communication, voice interruptions occur, affecting the quality of service to customers.

Therefore, some communication devices are provided with a function of monitoring RTP packets and detecting the occurrence of voice interruptions. A technique has been proposed in which, when a packet loss is detected in an IP telephone, voice data included in the lost packet is complemented (see, for example, Patent Document 1). The IP telephone device disclosed in Patent Document 1 detects the occurrence of packet loss based on sequence numbers indicating the order of received packets.

JP 2009-181050 A

However, the IP telephone device disclosed in Patent Document 1 can detect packet loss that occurs during a call, but cannot detect packet loss at the start or end of a call.

A packet loss detection system according to the present invention includes a network communication unit that transmits and receives packets including audio data to and from a communication destination via a network, converts an input audio signal into the packet, converts the received audio data into an audio signal, and converts the input audio signal into the packet. a voice communication unit that converts and outputs the voice data, extracts the voice data from the reception packet that is the packet received by the network communication unit from the network, transmits it to the voice communication unit, and receives it from the voice communication unit a transmission/reception unit configured to transmit the packet to the network via the network communication unit as a transmission packet; and the number of received packets, which is the number of received packets. If the number is less than the number of transmitted packets, it is determined that there is a loss of the received packets.

A packet loss detection method according to the present invention is a packet loss detection method for detecting loss of a packet containing audio data transmitted and received via a network, wherein the received packet, which is the packet received via the network, a step of extracting the audio data and transmitting the packets containing the input audio data to the network as transmission packets; a transmission packet number that is the number of transmission packets and a reception packet number that is the number of reception packets; and comparing the extracted number of transmitted packets and the number of received packets, and determining that there is a loss of the received packets when the number of received packets is smaller than the number of transmitted packets. ,

According to the present invention, the presence or absence of packet loss is determined by comparing the number of received packets and the number of transmitted packets. Therefore, it is possible to detect not only packet loss that occurs during a call, but also packet loss that occurs at the start and end of a call.

1 is a block diagram showing one configuration example of a communication system including the packet loss detection system of Embodiment 1; FIG. 2 is a diagram showing a configuration example of a packet transmitted and received via the IP communication network shown in FIG. 1; FIG. 3 is a diagram showing a configuration example of an RTP header attached to the packet shown in FIG. 2; FIG. 2 is a block diagram showing a configuration example of VoIP-GW 11 shown in FIG. 1; FIG. FIG. 4 is a diagram illustrating an example of sequence numbers of received packets when there is no packet loss; FIG. 10 is a diagram showing an example of sequence numbers of a plurality of received packets when packet loss occurs on the way; FIG. 10 is a diagram showing an example of sequence numbers when packet loss occurs at the start of a call; FIG. 10 is a diagram showing an example of sequence numbers when packet loss occurs just before the end of a call; 2 is a flow chart diagram showing an example of an operation procedure of VoIP-GW in Embodiment 1. FIG. FIG. 10 is a flow chart diagram showing an example of an operation procedure of VoIP-GW in Embodiment 2;

Embodiment 1.
A configuration of a communication system including the packet loss detection system of the first embodiment will be described. FIG. 1 is a block diagram showing a configuration example of a communication system including a packet loss detection system according to Embodiment 1. FIG.

Communication system 10 has VoIP-GW 11 , VoIP-GW 12 , telephone 21 , telephone 22 and SIP server 31 . VoIP-GW 11 , VoIP-GW 12 and SIP server 31 are connected to IP communication network 100 . Telephone 21 is connected to IP communication network 100 via VoIP-GW 11 . Telephone 22 is connected to IP communication network 100 via VoIP-GW 12 . The telephone 22 is a telephone used by a person with whom the user of the telephone 21 makes a call via the telephone 21 .

The IP communication network 100 is a network including a plurality of communication devices (not shown) that perform IP communication, and these communication devices are connected by wires and wirelessly. In the first embodiment, the network to which the VoIP-GW 11, VoIP-GW 12 and SIP server 31 are connected is the Internet, but the network is not limited to the Internet.

The VoIP-GW 11 is a gateway device that has a SIP-based VoIP function and relays communication between the telephone 21 and the IP communication network 100 . The VoIP-GW 12 has a SIP-based VoIP function and relays communication between the telephone 22 and the IP communication network 100 . The SIP server 31 is a server that implements a call function between the telephones 21 and 22 in the IP communication network 100 by call control using SIP.

The telephone 21 converts voice input via a microphone (not shown) from an analog signal to a digital signal and transmits it to the VoIP-GW 11, and converts the voice signal received from the VoIP-GW 11 from a digital signal to an analog signal. to output sound from a speaker (not shown). The telephone 22 converts voice input via a microphone (not shown) from an analog signal to a digital signal and transmits it to the VoIP-GW 12, and converts the voice signal received from the VoIP-GW 12 from a digital signal to an analog signal. to output sound from a speaker (not shown).

In the first embodiment, VoIP-GW 11 and VoIP-GW 12 have the same configuration, and the configuration of VoIP-GW 11 will be described in detail below, and detailed description of VoIP-GW 12 will be omitted. The VoIP-GW 12 is not limited to having the same configuration as the VoIP-GW 11, and may have a configuration different from that of the VoIP-GW 11. FIG.

Next, the configuration of packets transmitted and received between VoIP-GW 11 and VoIP-GW 12 via IP communication network 100 will be described. FIG. 2 is a diagram showing a configuration example of packets transmitted and received via the IP communication network shown in FIG. FIG. 3 is a diagram showing a configuration example of an RTP header attached to the packet shown in FIG.

As shown in FIG. 2, the packet 50 has a configuration in which an RTP header, a UDP (User Datagram Protocol) header, and an IP header are added to audio data. Although not shown in FIG. 2, the packet 50 may be provided with an error check code.

The IP header contains information on the destination IP address and the source IP address. A UDP header is attached to a connectionless communication packet. Since real-time performance is emphasized in voice communication, UDP is applied to packets 50 containing voice data rather than TCP (Transmission Control Protocol), which emphasizes reliability of communication. The UDP header contains parameter information such as source port number, destination port number, and total packet size.

The RTP header is added to data that requires real-time communication, such as audio and video. The RTP header, as shown in FIG. 3, contains sequence numbers indicating the order of packets to be transmitted and time stamp information for enabling real-time playback. Besides these information, the RTP header contains information of parameters such as version number (V), payload type (PT) and synchronization source identifier. The payload type is information indicating the type of code (encoding) of audio data to be transmitted.

Generally, when the packet 50 shown in FIG. 2 is transmitted and received between two communication devices via the IP communication network 100, the packet 50 is processed in a plurality of hierarchies inside the two communication devices. are processed in order. For example, consider a case where multiple layers are layered in order from the lowest layer, the network layer, to the transport layer, the session layer, and the highest layer, the application layer.

Of the two communication devices, the source communication device first divides an audio signal in the application layer to generate packets containing audio data. Next, the transmission source communication device adds an RTP header to the audio data packet in the session layer. Subsequently, the source communication device adds a UDP header to the RTP header-added packet in the transport layer. Furthermore, the transmission source communication device generates a packet 50 by adding an IP header to the packet to which the UDP header is added in the network layer. Then, the source communication device transmits the generated packet 50 to the destination communication device via the IP communication network 100 .

On the other hand, when the destination communication device receives the packet 50 from the transmission source communication device, the network layer reads the IP header information, removes the IP header from the packet 50, and processes the upper transport layer. . In the transport layer, the destination communication device reads the UDP header information, removes the UDP header from the packet, and performs session layer processing in the upper layer. In the session layer, the destination communication device reads the RTP header information, removes the RTP header from the packet, and performs processing in the highest application layer. Then, the destination communication device converts the audio data extracted from the packet 50 into an audio signal in the application layer, connects the audio data to the already converted audio signal, and outputs the audio signal.

In this manner, the packet 50 is generated by processing the audio data in order from the upper layer to the lower layer in the transmission source communication device, and the packet 50 is generated in order from the lower layer to the upper layer in the transmission destination communication device. It is restored to audio data by being processed. Hereinafter, communication using the packet 50 to which the RTP header is attached is referred to as RTP communication.

Next, the configuration of VoIP-GW 11 shown in FIG. 1 will be described with reference to FIG. FIG. 4 is a block diagram showing one configuration example of the VoIP-GW 11 shown in FIG. Here, mainly the case where VoIP-GW 11 communicates with VoIP-GW 12 will be described. As shown in FIG. 4, the VoIP-GW 11 includes an IP communication interface 1, an IP communication unit 2, a SIP communication unit 3, a voice communication unit 4, a telephone interface 5, a transmission/reception unit 6, and a voice quality monitoring unit 7. and

The IP communication interface 1 is a device that serves as a communication interface for performing IP communication, and is connected to the IP communication network 100 . When the IP communication unit 2 transmits a packet received from the SIP communication unit 3 or the transmission/reception unit 6 to the IP communication network 100 via the IP communication interface 1, the IP communication unit 2 adds an IP header and a UDP header or a TCP header to the packet. to send. When the IP communication unit 2 receives the packet 50 from the IP communication network 100 via the IP communication interface 1, the IP communication unit 2 reads the IP header and the UDP header or TCP header of the packet 50, removes these headers, and sends the packet to the SIP communication unit. 3 or the transmission/reception unit 6.

Further, IP communication unit 2 performs IP communication for transmitting and receiving packets to and from VoIP-GW 12 or SIP server 31 via IP communication interface 1 in accordance with TCP/IP during call control between telephone set 21 and telephone set 22 . When transmitting/receiving the packet 50 to/from the VoIP-GW 12, the IP communication unit 2 performs IP communication of transmitting/receiving the packet 50 to/from the VoIP-GW 12 via the IP communication interface 1 according to UDP/IP. The IP communication section 2 corresponds to the network communication section in the present invention.

The SIP communication unit 3 communicates via the IP communication unit 2 according to SIP. When establishing a SIP session, the SIP communication unit 3 communicates with the SIP server 31 to determine RTP parameters to be used for voice communication, and notifies the transmission/reception unit 6 of information on the determined parameters. Also, the SIP communication unit 3 performs call control at the time of starting and ending a call performed via the telephone 21 and the telephone 22 . In the first embodiment, the communication object between the telephone 21 and the telephone 22 is voice. Determine the RTP parameters to be used.

Transmitting/receiving unit 6 reads RTP header information from a received packet, which is a packet received from VoIP-GW 12 of the communication destination via IP communication network 100 , extracts voice data from the received packet, and passes the extracted voice data to voice communication unit 4 . Further, the transmitting/receiving unit 6 converts voice data received from the voice communication unit 4 into a transmission packet, which is a packet to be transmitted to the VoIP-GW 12 of the communication destination. At that time, the transmission/reception unit 6 adds an RTP header in which parameter information determined by the SIP communication unit 3 is written to the transmission packet. The transmitter/receiver 6 counts the number of transmitted packets and the number of received packets from the start to the end of call control between the telephones 21 and 22 . In the following description, the number of transmitted packets is CT, and the number of received packets is CR.

Upon receiving a voice signal from the telephone interface 5 , the voice communication unit 4 converts the voice signal into voice data and transmits a packet containing the voice data to the transmission/reception unit 6 . Upon receiving a packet from the transmission/reception unit 6 , the voice communication unit 4 extracts voice data from the packet, converts the extracted voice data into a voice signal, and transmits the voice signal to the telephone interface 5 . The telephone interface 5 is a device that serves as an interface for transmitting and receiving voice signals to and from the telephone 21 and is connected to the telephone 21 .

Voice quality monitoring unit 7 includes determination unit 8 . The voice quality monitoring unit 7 acquires statistical information about communication of packets 50 related to voice communication from the voice communication unit 4, and detects problems related to quality such as the occurrence of voice interruptions. When detecting a quality problem, the voice quality monitoring unit 7 notifies the user of the detected quality problem by means of a log, an alarm, or the like.

The determination unit 8 acquires the transmission packet number CT and the reception packet number CR extracted by the transmission/reception unit 6 from the transmission/reception unit 6, and determines whether or not the transmission packet number CT and the reception packet number CR are equal. If the number of received packets CR is smaller than the number of transmitted packets CT, the determination unit 8 determines that the received packets have been lost. determine that it is not. This is because a call is generally made bidirectionally between the user of the telephone 21 and the user of the telephone 22, so that transmission and reception of data communication in an RTP call are performed at the same time, and the number of transmitted packets and the number of received packets are equal. be. In the case of the first embodiment, the number of packets transmitted from the VoIP-GW 12 to the VoIP-GW 11 is equal to the received packet number CR while the users are talking to each other using the telephones 21 and 22 .

An example of the hardware configuration of the VoIP-GW 11 shown in FIG. 4 will now be described. The VoIP-GW 11 has a memory and a CPU (Central Processing Unit) not shown. The IP communication unit 2, the SIP communication unit 3, the voice communication unit 4, the voice quality monitoring unit 7, and the transmission/reception unit 6 are configured by the CPU executing the programs stored in the memory. Alternatively, part or all of the IP communication unit 2, SIP communication unit 3, voice communication unit 4, voice quality monitoring unit 7, and transmission/reception unit 6 may be configured by a dedicated circuit such as ASIC (Application Specific Integrated Circuit).

In Embodiment 1, a case where the packet loss loss system of the present invention is configured in VoIP-GW 11 will be described, but some of the functions of VoIP-GW 11 described above are performed by other devices, such as telephone 21 or IP It may be provided in a communication device (not shown) such as a router in the communication network 100 . Next, before explaining the operation of the VoIP-GW 11 of the first embodiment, the operation of the gateway device of the comparative example will be explained.

Here, the gateway device (not shown) of the comparative example detects the occurrence of packet loss based on the sequence number indicating the order of transmitted packets, like the IP telephone device disclosed in Patent Document 1. It shall have the function. In this case, the gateway device of the comparative example can detect packet loss that occurs during a call, but cannot detect packet loss at the start and end of a call. This will be explained concretely.

As described with reference to FIGS. 2 and 3, RTP communication is performed using UDP because real-time performance is emphasized. Since UDP is a connectionless communication, packet loss cannot be detected in transport layer and session layer processing, and packet loss must be detected in layers higher than these layers. Therefore, the gateway device of the comparative example detects packet loss by monitoring the sequence number written in the RTP header in the session layer.

A sequence number is a consecutive number assigned to a transmitted packet, and a number that increases by one is assigned for each transmission of a packet. The gateway device of the comparative example can detect that a packet loss has occurred if the sequence numbers are discontinuous when sequentially reading the sequence numbers of the received packets. Also, the gateway device of the comparative example can confirm how much voice interruption has occurred by counting the number of packets with discontinuous sequence numbers.

FIG. 5 is a diagram illustrating an example of sequence numbers of received packets when there is no packet loss. As shown in FIG. 5, the sequence numbers of a series of received packets are consecutive. In this case, after receiving 10 packets, it can be seen that there is no packet loss and that there is no problem with the voice quality. The gateway device of the comparative example determines that the number of receptions=10 and the number of losses=0.

FIG. 6 is a diagram showing an example of sequence numbers of a plurality of received packets when packet loss occurs on the way. As shown in FIG. 6, the sequence numbers of multiple received packets are not continuous. In the example shown in FIG. 6, it can be seen that sequence numbers 5 to 7 are missing in the middle, three packets are lost, and the audio is interrupted. The gateway device of the comparative example determines that the number of receptions=10 and the number of losses=3.

However, the gateway device of the comparative example can detect packet loss only when the middle of the sequence number is missing as shown in FIG. Two specific examples of packet loss that cannot be detected by the gateway device of the comparative example will be described below.

The first is when packet loss occurs immediately after the start of a call. FIG. 7 is a diagram showing an example of sequence numbers when packet loss occurs at the start of a call. From the security point of view of avoiding attacks such as spoofing due to guessing of the sequence number, the initial sequence number at the start of voice communication is random. Since RTP does not have a mechanism for negotiating the sequence number at the start of communication, the destination communication device cannot know in advance from what number the sequence number will start.

Therefore, if packet loss occurs immediately after the start of a call, it is not possible to distinguish whether the sequence number of the first received packet is the normal leading number or not, just by looking at the sequence number. Loss cannot be detected. The example shown in FIG. 7 shows a state in which five packets with sequence numbers 1 to 5 have not been received after the start of a call. In the example shown in FIG. 7, the gateway device of the comparative example determines that the number of receptions=5 and the number of losses=0. In this case, the gateway device of the comparative example cannot obtain the number of lost packets.

The second is when packet loss occurs when the call ends. FIG. 8 is a diagram showing an example of sequence numbers when packet loss occurs just before the end of a call. FIG. 8 shows a state in which six packets with sequence numbers 5 to 10 were not received immediately before the end of the call. In the case of the example shown in FIG. 8, the communication is terminated while packet loss is occurring, and the communication is cut off without being able to receive the packet with the next sequence number. I can't put it on. Therefore, the gateway device of the comparative example determines that the number of receptions=4 and the number of losses=0. Also in this case, the gateway device of the comparative example cannot obtain the number of lost packets.

So far, the operation of the gateway device of the comparative example has been described. Next, operations of the communication system 10 and operations of the VoIP-GW 11 according to the first embodiment will be described. Here, the case where the user of the telephone 21 goes off-hook and makes an outgoing call request from the telephone 21 to the telephone 22 will be described, but the telephone 22 may issue an incoming call request to the telephone 21 . FIG. 9 is a flow chart diagram showing an example of the operation procedure of the VoIP-GW according to the first embodiment.

When there is a call request from the telephone 21 to the VoIP-GW 11, the SIP communication unit 3 of the VoIP-GW 11 transmits and receives information necessary for call control with the SIP server 31 and VoIP-GW 12, and according to SIP, VoIP-GW 12 and establish a SIP session. At that time, the SIP communication unit 3 communicates with the SIP server 31 various parameters (source IP address, source port number, destination IP address and destination port number) used for voice communication of the call. is also negotiated. The SIP communication unit 3 notifies the transmission/reception unit 6 of the RTP parameters determined with the SIP server 31 .

When RTP communication becomes possible between VoIP-GW 11 and VoIP-GW 12, VoIP-GW 11 converts voice signals received from telephone 21 into packets 50 and transmits them to VoIP-GW 12. FIG. VoIP-GW 12 converts packet 50 received from VoIP-GW 11 into a voice signal and transmits the voice signal to telephone 22 . VoIP-GW 12 converts voice signals received from telephone 22 into packets 50 and transmits the packets to VoIP-GW 11 . VoIP-GW 11 converts packet 50 received from VoIP-GW 12 into a voice signal and transmits the voice signal to telephone 21 . In this manner, a voice call is made between the telephones 21 and 22. FIG. After that, when the user of the telephone 21 or 22 goes on-hook, the call control between the VoIP-GW 11 and the VoIP-GW 12 ends, and the voice call ends.

During voice communication between the telephones 21 and 22, the transmission/reception unit 6 counts and holds the number of transmitted packets CT and the number of received packets CR. When the voice call ends, in step S201 of FIG. 9, the transmitting/receiving section 6 extracts the number of transmitted packets CT and the number of received packets CR. Then, the determination unit 8 acquires information on the number of transmitted packets CT and the number of received packets CR from the transmitter/receiver 6, and determines whether or not the number of received packets CR is smaller than the number of transmitted packets CT (step S202).

If the received packet count CR is equal to the transmitted packet count CT, the determining unit 8 determines that there is no received packet loss (step S203). On the other hand, if the number of received packets CR is smaller than the number of transmitted packets CT as a result of the determination processing in step S202, the determination unit 8 determines that there is a loss of received packets (step S204). In this case, assuming that the number of received packets lost is CLL, the number of received packets lost CLL may be calculated using the formula CLL=CT-CR.

When the determination unit 8 determines that there is a received packet loss, the voice quality monitoring unit 7 checks the received packet loss number CLL, and notifies the user of the telephone 21 of the occurrence of voice interruption through a log or an alarm. do. The user may be notified by voice through a speaker (not shown) of the telephone 21. If the telephone 21 is provided with a display (not shown), a message may be displayed on the display. may notify you.

The packet loss detection system of the first embodiment has an IP communication section 2, a voice communication section 4, a transmission/reception section 6, and a determination section 8. FIG. IP communication unit 2 transmits and receives packet 50 including voice data to and from a communication destination via IP communication network 100 . The voice communication unit 4 converts voice signals input from the telephone 21 into packets containing voice data, converts voice data received from the transmission/reception unit 6 into voice signals, and outputs the voice signals to the telephone 21 . Transmitting/receiving unit 6 extracts voice data from a received packet received by IP communication unit 2 from IP communication network 100, transmits the voice data to voice communication unit 4, and transmits the packet received from voice communication unit 4 to the IP communication unit as a transmission packet. 2 to the IP communication network 100 . The determination unit 8 determines whether or not there is a loss in the received packet. The transmitting/receiving unit 6 extracts the number of transmitted packets CT and the number of received packets CR. The determining unit 8 compares the number of transmitted packets CT and the number of received packets CR extracted by the transmitting/receiving unit 6, and determines that there is a loss of received packets when the number of received packets CR is smaller than the number of transmitted packets CT.

According to the first embodiment, when monitoring voice quality, the presence or absence of packet loss is determined by comparing the number of received packets CR and the number of transmitted packets CT. Therefore, packet loss can be determined without using the sequence number written in the RTP header, so not only packet loss that occurs during a call but also packet loss that occurs at the start and end of a call can be detected. be able to. According to the first embodiment, it is possible to detect packet loss that occurs at the start and end of a call by using information on the number of packets within the range of existing protocols, and to analyze special voice data. The functionality does not have to be implemented in the communication device.

Embodiment 2.
In the second embodiment, the sequence number is also used to detect the occurrence of packet loss. In the second embodiment, the same components as those in the first embodiment are denoted by the same reference numerals, and detailed description thereof will be omitted. Further, in the second embodiment, operations different from those in the first embodiment will be described in detail, and detailed descriptions of operations similar to those in the first embodiment will be omitted.

The configuration of the VoIP-GW 11 in the communication system 10 of Embodiment 2 will be explained with reference to FIG. In the second embodiment, the transmitting/receiving section 6 reads the information of the sequence number of the RTP header of the received packet and checks the missing sequence number during the call, in addition to the operation described in the first embodiment. Then, the transmitting/receiving unit 6 counts and holds the number of received packets lost, which is the number of received packets lost during a call, from the missing sequence number. In the following, the number of received packet losses is defined as CL.

Also in the second embodiment, a case where the packet loss loss system of the present invention is configured in the VoIP-GW 11 will be described, but some of the functions of the VoIP-GW 11 may be replaced by other devices such as the telephone 21 or IP It may be provided in a communication device (not shown) such as a router in the communication network 100 .

Next, the operation of VoIP-GW 11 of the second embodiment will be explained with reference to FIG. FIG. 10 is a flow chart diagram showing an example of a VoIP-GW operation procedure according to the second embodiment. In Embodiment 2, the call control by VoIP-GW 11, VoIP-GW 12 and SIP server 31 is the same as the control described in Embodiment 1, so detailed description thereof will be omitted.

During voice communication between the telephones 21 and 22, the transmission/reception unit 6 counts and holds the number of transmitted packets CT and the number of received packets CR. Further, based on the sequence numbers of the received packets, the transmitting/receiving unit 6 counts and holds the received packet loss count CL corresponding to the missing sequence numbers. When the voice call ends, in step S301 of FIG. 10, the transmitting/receiving section 6 extracts the number of transmitted packets CT and the number of received packets CR.

In step S302, the transmitting/receiving unit 6 extracts the received packet loss count CL corresponding to the missing sequence number based on the sequence number of the received packet. Then, the determination unit 8 acquires information on the transmission packet count CT, the reception packet count CR, and the reception packet loss count CL from the transmission/reception unit 6 . Further, the determination unit 8 determines whether or not the sum of the received packet count CR and the received packet loss count CL is smaller than the transmitted packet count CT (step S303).

If the sum of the received packet count CR and the received packet loss count CL is equal to the transmitted packet count CT, the determining unit 8 determines that there is no received packet loss at the start and end of the call (step S304). On the other hand, if the sum of the received packet count CR and the received packet loss count CL is smaller than the transmitted packet count CT as a result of the determination processing in step S302, the determination unit 8 determines that there is no received packet loss at least at the start or end of the call. It is determined that there is (step S305). In this case, if the number of received packet losses at one or both of the start and end of a call is CLL, the determination unit 8 calculates the number of received packet losses CLL using the formula CLL=CT-(CR+CL). You may

When the determining unit 8 determines that there is a loss of received packets, the voice quality monitoring unit 7 checks the number of received packet losses CL and CLL, and notifies the user of the telephone 21 by a log or an alarm that voice interruption has occurred. to notify you. The user may be notified by voice through a speaker (not shown) of the telephone 21. If the telephone 21 is provided with a display (not shown), a message may be displayed on the display. may notify you.

Effects of the second embodiment will be described. If the relationship is "the number of received packets + the number of lost packets received during a call<the number of transmitted packets", packets are lost at one or both of the start and end of a call. In the prior art, packet loss can be detected in the middle of a call, so if the second embodiment is applied, it is possible to know when packet loss occurs in the middle of a call, or at the start or end of a call. become. Since there are multiple communication devices such as routers and gateway devices on the telephone path, the user's declaration (declaration of when the sound is interrupted during a call) and the detection results of each communication device (this implementation By comparing the detection result of whether packet loss occurred in the middle of a call or at the start or end of a call detected using the method of form 2, it is possible to determine which communication has a packet loss that matches the user's report. It will be a clue to identify whether it occurred in the device.

For example, when a user reports to an IP telephone company that provides an IP telephone service that the voice is interrupted at the start of a call, the person in charge of the IP telephone company identifies in which communication device the abnormality occurred. It is conceivable that you want to In this case, if there is a communication device that detects packet loss in the middle of a call and a communication device that detects packet loss at the start or end of a call among the plurality of communication devices, the user will report it. Anomalies can be identified as occurring in a communication device where packet loss is detected at the beginning or end of a call.

The present invention is not limited to the first and second embodiments described above, and can be applied to other usage forms as follows.
(Application example 1)
In the first and second embodiments, the RTP communication is used for voice communication, but it can also be applied to video communication to detect video breaks.
(Application example 2)
In the first and second embodiments, detection of packet loss is performed as detection of voice interruption, but the present invention can also be applied to network communication quality inspection.
(Application example 3)
In Embodiments 1 and 2, the case where the transmission/reception unit 6 counts the number of communication packets has been described, but other communication function units such as a packet high-speed transfer module may perform the counting.
(Application example 4)
In Embodiments 1 and 2, the case of application to RTP communication has been described, but other protocols can also be applied as long as they are communications in which transmission and reception are performed simultaneously.

1 IP communication interface 2 IP communication unit 3 SIP communication unit 4 Voice communication unit 5 Telephone interface 6 Transmission/reception unit 7 Voice quality monitoring unit 8 Judging unit 10 Communication system 11 VoIP-GW
12 VoIP-GW
21 telephone 22 telephone 31 SIP server 50 packet 100 IP communication network

Claims (3)

a network communication unit that transmits and receives packets containing audio data to and from a communication destination via a network;
an audio communication unit that converts an input audio signal into the packet, converts received audio data into an audio signal, and outputs the audio signal;
extracting the audio data from the received packet, which is the packet received by the network communication unit from the network, and transmitting the audio data to the audio communication unit; and using the packet received from the audio communication unit as a transmission packet, the network communication unit a transmitting/receiving unit that transmits to the network via
a determination unit that determines whether or not there is a loss in the received packet;
The transmitting/receiving unit extracts the number of transmitted packets, which is the number of transmitted packets, and the number of received packets, which is the number of received packets;
The determining unit compares the number of transmitted packets extracted by the transmitting/receiving unit with the number of received packets, and determines that there is a loss of the received packets when the number of received packets is smaller than the number of transmitted packets.
Packet loss detection system. the packets received by the network communication unit are assigned consecutive sequence numbers indicating the order in which they were transmitted;
The transmitting/receiving unit extracts a received packet loss count, which is the number of received packets corresponding to the missing sequence number, based on the sequence numbers of the received packets,
The determination unit determines that there is a packet loss before and after the received packet when the sum of the number of received packets and the number of received packet losses is smaller than the number of transmitted packets.
The packet loss detection system according to claim 1. A packet loss detection method for detecting loss of packets containing audio data transmitted and received via a network,
a step of extracting the voice data from a received packet, which is the packet received via the network, and transmitting the packet containing the input voice data to the network as a transmission packet;
extracting the number of transmitted packets, which is the number of transmitted packets, and the number of received packets, which is the number of received packets;
comparing the extracted number of transmitted packets and the number of received packets, and determining that there is a loss of the received packets when the number of received packets is smaller than the number of transmitted packets;
A packet loss detection method comprising:

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