JP2005175694A - Voice communication apparatus and voice communication method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To realize a call with a relatively good sound quality from the start to the end of the call even when a network congestion state changes with time when the call is realized by a voice packet.
SOLUTION: Receiving means for receiving a voice packet including voice data encoded by a transmission side codec via a network, and evaluating a sound quality of a voice reproduced by decoding the received voice packet by a reception side codec. An audio communication apparatus comprising: an evaluation unit that performs the change and a change unit that changes the codec on each of the transmission side and the reception side based on the evaluation result. For example, when it is evaluated that the sound quality has deteriorated, the changing means changes the codec on each of the transmission side and the reception side to a codec having a bit rate lower than the bit rate.
[Selection] Figure 2

Description

  In the case where the present invention realizes a call using voice packets, even if the network congestion status changes over time (due to sound quality deterioration or subsequent sound quality recovery, etc.) The present invention relates to a technology for realizing a call with good sound quality.

In recent years, VoIP call systems have attracted attention, and various technical proposals have been made (for example, see Patent Document 1).
JP 2000-31229 A

  However, in VoIP, the codec used for voice encoding and decoding is selected at the start of communication based on the capabilities and priority of both transmission and reception. Even if the sound quality deteriorates due to lack of communication or a change in communication status during communication, no response was possible. Therefore, there is a problem that it is difficult to make a call with good sound quality from the start to the end of the call.

  The problem of the present invention is that when a call is realized by a voice packet (for example, a VoIP packet), the congestion status of the network (for example, an IP network) has changed over time (due to a decrease in sound quality or a subsequent recovery of sound quality). Another object is to provide a technique for realizing a call with relatively good sound quality from the start to the end of the call.

  The present invention has been made in order to solve the above-described problem, and is a voice communication apparatus, and a receiving means for receiving voice packets including voice data encoded by a transmission-side codec via a network, Evaluation means for evaluating the sound quality of the voice to be reproduced by decoding the received voice packet by the reception side codec, and changing means for changing the codec of each of the transmission side and the reception side based on the evaluation result The configuration.

  According to the present invention, it is possible to evaluate (or estimate) the sound quality of speech on the receiving side, and change the codec on each of the transmitting side and the receiving side based on the evaluation result (or estimation result). That is, when a call is realized by a voice packet (for example, a VoIP packet), even if the congestion status of the network (for example, an IP network) changes over time (sound quality deterioration or subsequent sound quality recovery, etc.) From the start to the end, it is possible to realize a call with relatively good sound quality.

  Further, in the voice communication apparatus, for example, when it is evaluated that the sound quality is deteriorated, the changing unit changes the codec of each of the transmitting side and the receiving side to a codec having a bit rate lower than the bit rate. . This shows an example of codec change by the changing means.

  In this way, in a case where a call is realized by a voice packet (for example, a VoIP packet), even if the congestion status of the network (for example, an IP network) changes over time (due to a decrease in sound quality), from the start to the end of the call. Through this, it is possible to realize a call with relatively good sound quality.

  Further, in the voice communication device, for example, when the changing means is evaluated that the sound quality has been recovered after it is evaluated that the sound quality has deteriorated, the changed codec on each of the transmitting side and the receiving side is changed. In addition, the codec is changed to a bit rate higher than that of the codec. This also shows an example of codec change by the changing means.

  In this way, when a call is realized by a voice packet (for example, a VoIP packet), even if the congestion status of the network (for example, an IP network) changes over time (sound quality recovery after sound quality deterioration), From the start to the end, it is possible to realize a call with relatively good sound quality.

The present invention can also be specified as a method invention as follows.
A reception step of receiving voice packets including voice data encoded by the transmission side codec via the network; an evaluation step of evaluating the sound quality of the voice reproduced by decoding the received voice packets by the reception side codec; and And a changing step of changing the codec of each of the transmission side and the reception side based on the evaluation result.

  In the voice communication method, for example, when it is evaluated that the sound quality has deteriorated, the changing step changes the codec on the transmitting side and the receiving side to a codec having a bit rate lower than the bit rate.

  Further, in the voice communication method, for example, when the change step is evaluated that the sound quality has been recovered after it is evaluated that the sound quality has deteriorated, the codec after the change on each of the transmitting side and the receiving side is changed. In addition, the codec is changed to a bit rate higher than that of the codec.

  According to the present invention, when a call is realized by a voice packet (for example, a VoIP packet), it is assumed that the congestion state of the network (for example, an IP network) has changed over time (due to a decrease in sound quality or a subsequent recovery of sound quality). However, it is possible to realize a call with relatively good sound quality from the start to the end of the call.

Hereinafter, a communication system including a voice communication apparatus according to an embodiment of the present invention will be described with reference to the drawings. FIG. 1 is a functional block diagram of a voice communication apparatus according to an embodiment of the present invention.
(Configuration of voice communication device)
The voice communication device 100 is a communication device having a function of realizing a voice call on an IP network by VoIP (Voice over IP). For example, when configuring a phone-to-phone type IP telephone in which an existing general telephone is connected to an IP network via a modem with a VoIP function, the voice communication apparatus 100 is connected to the modem with the VoIP function. It corresponds to. Further, when configuring a PC to phone type IP phone connected to an IP network via an information processing device such as a general personal computer, the information processing device such as the personal computer corresponds to the voice communication device 100. To do. Further, when configuring an IP telephone with a built-in VoIP function, the telephone itself corresponds to the voice communication apparatus 100.

As shown in FIG. 1, a voice communication apparatus 100 includes a codec group (a plurality of codecs) 101, a communication function 102, a communication state acquisition function 103, a sound quality evaluation function 104, a jitter buffer 105, a decoding function 106, a voice An output function 107, a codec information storage / provision function 108, a codec selection / change request function 109, a codec change request response function 110, a voice input function 111, an encode function 112, a calling / incoming call function, and the like. Yes. These functions are realized by predetermined software or a circuit.

The voice communication device 100 is connected to the IP network via the communication function 102. The decoding function 106 and the encoding function 112 are functions realized by a selected or changed codec, as will be described later. Note that the codec 101 is an encoding / decoding device or device used for digitizing audio that is an analog signal. The jitter buffer 105 is a buffer provided in anticipation of delay and fluctuations for reproduction at equal intervals. If a delay or fluctuation exceeding the buffer value occurs, the packet is discarded. The codec information storing / providing function 108 stores the correspondence relationship between the codec 101 and codec information such as the bit rate, and provides the sound quality evaluation function 104 with codec information corresponding to the currently used codec. Is. Other functions will be clarified by the following operation description.
(Operation of voice communication device)
Next, the operation of the voice communication apparatus 100 configured as described above will be described with reference to the drawings. FIG. 2 is a flowchart for explaining the operation of the voice communication apparatus 100. Hereinafter, for convenience of explanation, one voice communication device 100 is focused on its transmission function as a transmitting voice communication device (hereinafter referred to as voice communication device 100T), and the other voice communication device 100 is used as its reception function. A case will be described as an example where the receiving side voice communication apparatus (hereinafter referred to as voice communication apparatus 100R) performs voice communication (for example, a telephone call) between the two. Actually, since the voice communication devices 100T and 100R have both the transmission function and the reception function, the voice communication devices 100T and 100R perform transmission and reception simultaneously in parallel. Become.
(Outline of codec change processing)
As shown in FIG. 2, first, communication connection processing is performed between the voice communication apparatuses 100T and 100R (S100). An example of such processing is codec selection processing. This codec selection process is a process for selecting an appropriate codec from the plurality of codecs 101 in accordance with the congestion status of the IP network in each of the voice communication apparatuses 100T and 100R. The codecs selected in this way are called a transmission side codec and a reception side codec. As a result, an appropriate codec is selected in each of the voice communication apparatuses 100T and 100R, and good voice communication can be performed between the two. This is an existing process.

  Next, the voice communication device 100R transmits a voice packet (voice data including voice data input via the voice input function 111 and encoded by the transmission side codec (encoding function 112)) transmitted from the voice communication device 100T on the transmission side. When a packet arrives via the IP network (S101), it receives this (S101). This is performed mainly by the communication function 102 (corresponding to the receiving means of the present invention), for example. At the same time, the voice communication device 100R obtains the arrival state of the voice data included in the payload portion of the received voice packet (S102). This is performed mainly by the communication state acquisition function 103, for example. The voice communication device 100R then evaluates the sound quality (S103, S104). Specifically, the received voice packet is decoded by the receiving side codec and the sound quality of the reproduced voice is evaluated (or estimated). This sound quality evaluation is performed mainly by the sound quality evaluation function 104 (corresponding to the evaluation means of the present invention). Details of the sound quality evaluation will be described later.

Based on the evaluation result (S104, S106), the voice communication device 100R changes the codec on each of the transmission side and the reception side, that is, the codec currently used by the voice communication device 100T on the transmission side and each of the voice communication device 100R ( S105, S107 to S114).
(Details of codec change processing)
Details of the codec change processing will be described. The following processing is performed mainly by the codec selection / change request function 109, for example.
(Codec change processing when sound quality deteriorates)
First, the codec change process when the sound quality is degraded will be described. As a result of the sound quality evaluation, if it is determined that the sound quality has deteriorated (S104: bad), the voice communication device 100R selects a codec (reception side codec) (S105). Details of the codec selection will be described later.

  When the codec is selected, the voice communication device 100R transmits a change request including specific data for specifying the selected codec (for example, in the form of a data packet) to the voice communication device 100T on the transmission side ( S108). This is to change the transmission side codec to the codec specified by the specific data.

  When the change request transmitted from the receiving voice communication apparatus 100R arrives via the IP network (S108), the transmitting voice communication apparatus 100T receives the change request. This reception is performed mainly by the communication function 102, for example. Then, the voice communication device 100T determines whether or not the codec specified by the specific data included in the change request can be used by itself (S109). When it is determined that the codec is usable (S109: Possible), the voice communication device 100T returns a “change” (for example, in the form of a data packet) to the voice communication device 100R (S110).

  When the “change” transmitted from the transmitting voice transmitting device 100T arrives via the IP network (S111), the voice communication device 100R receives this and sends a confirmation response to the voice communication device 100T. Return (S112). At the same time, the voice communication device 100R changes the currently used codec to the codec selected in S105. The confirmation response and the change are performed mainly by the function 110 for responding to the codec change request, for example.

  On the other hand, when the confirmation response transmitted from the receiving voice transmitting device 100R arrives via the IP network (S115), the voice communication device 100T determines the currently used codec based on the specific data received in S108 earlier. Change to a specific codec. Thereafter, a voice call is performed between the two using the changed codec.

  If it is determined in S109 that the voice communication device 100T cannot be used (S109: disabled), the voice communication device 100T returns a response “unavailable” to the voice communication device 100R (S113). In this case, the voice communication device 100R receives this and selects another codec (S114, S105). Then, as described above, the voice communication device 100R transmits a change request including specific data for specifying the selected codec (for example, in the form of a data packet) to the voice communication device 100T on the transmission side ( S108). By repeating this process, it is finally possible to change to a codec common to both. Although it may be assumed that the codec specified by the specific data does not exist in the voice communication apparatus 100T on the transmission side, in this case, it is conceivable to continue the voice communication (call) with different codecs between them. .

As described above, the sound quality is evaluated (or estimated) (S103), and the codec on each of the transmitting side and the receiving side is changed based on the evaluation result (S104 to S115). That is, here, even if the congestion status of the IP network changes over time (due to a decrease in sound quality), the voice communication apparatuses 100T and 100R are changed to appropriate codecs corresponding to the congestion status of the IP network. It will be. Therefore, even if the IP network congestion status changes over time (due to sound quality degradation), good voice communication (call) can be performed between the two parties from the start to the end of the voice communication.
(Codec change processing when sound quality is restored after sound quality is degraded)
Next, a description will be given of the codec change process when the sound quality is recovered after the sound quality is lowered as described above (S104: bad). As a result of the sound quality evaluation, if it is determined that the sound quality is improved (S104: good), the voice communication device 100R evaluates whether the sound quality of the currently used codec is worse than that at the start of communication (S106). As a result, when it is evaluated that the sound quality is poor (S106: bad), the voice communication device 100R selects a codec (S105). Details of the codec selection will be described later. As for the subsequent processes, the processes of S107 to S115 are performed in the same manner as described above.

  As described above, the sound quality is evaluated (or estimated) (S103), and the codec on each of the transmitting side and the receiving side is changed based on the evaluation result (S104 to S115). That is, here, even if the congestion status of the IP network changes over time (the sound quality is restored after the deterioration of the sound quality), an appropriate codec corresponding to the congestion status of the IP network is used in each of the voice communication apparatuses 100T and 100R. Will be changed. Therefore, even if the congestion status of the IP network changes over time (after the sound quality is lowered and then recovered), it is possible to perform good voice communication (call) between the two parties from the start to the end of the voice communication. Become.

(Details of sound quality evaluation part 1)
Next, details of the sound quality evaluation (S103) will be described with reference to the drawings. FIG. 3 is a flowchart for explaining the details of the sound quality evaluation. This is an example of a sound quality evaluation method, and shows processing performed by the voice communication device 100R on the receiving side.

  When the VoIP packet (voice packet including voice data encoded by the transmission side codec) transmitted from the transmission side voice communication apparatus 100T arrives via the IP network, the voice communication apparatus 100R receives this (S200). ), Data is input to the decoding function 106 via the jitter buffer 105. The input data is decoded by the currently used codec (for example, the receiving side codec selected in S101) (decoding function 106), and is output from the audio output function 107 as audio.

  The voice communication device 100R acquires the arrival time and size of the data input to the decoding function 106, and inputs this to the sound quality evaluation function 106 (S201). This is performed mainly by the communication state acquisition function 103, for example.

  The sound quality evaluation function 104 calculates the arrival interval of data (calculated from the arrival time input thereto) and information on the receiving side codec currently used (provided from the codec information storage / providing function 108). Compare (S202). Furthermore, the voice evaluation function 104 also calculates a bit rate (only the voice payload part) within a certain time (S203). Then, the voice evaluation function 104 has a problem in sound quality when any of the following conditions is satisfied, that is, the sound quality (the sound quality of the sound reproduced by decoding the voice packet received in S201 by the receiving side codec) It is evaluated (or estimated) that it has become worse (S204).

  (Condition 1) The data arrival interval continues to be longer than a certain length compared to the processing interval of the currently used receiving codec. (Condition 2) The bit rate within a certain period of time is lower than a certain value compared to the bit rate of the receiving codec currently used. Note that the bit rate of the codec indicates the amount of data recorded per second when the audio is digitized (encoded or decoded). The larger this, the higher the sound quality.

As described above, it is possible to perform sound quality evaluation (or estimation).
(Method 2 of sound quality evaluation)
Next, another example of the sound quality evaluation (S103) will be described with reference to the drawings. FIG. 4 is a flowchart for explaining another example of details of sound quality evaluation. This is also an example of the sound quality evaluation method, and shows the processing performed by the voice communication device 100R on the receiving side.

  When the VoIP packet (voice packet including voice data encoded by the transmission side codec) transmitted from the voice communication device 100T on the transmission side arrives via the IP network, the voice communication device 100R receives this (S300). ). The voice communication device 100R acquires the arrival time and size of the received VoIP packet and inputs them to the sound quality evaluation function 104 (S301). This is performed mainly by the communication state acquisition function 103, for example.

  The sound quality evaluation function 104 acquires a sequence number from the header (RTP header) of the received VoIP packet (S302). The sound quality evaluation function 104 is provided from the arrival interval of the VoIP packet (calculated from the arrival time input thereto) and information of the currently used receiving codec (codec information storing / providing function 108). (S303). Further, the sound quality evaluation function 104 also calculates a bit rate within a certain time (for example, only the voice payload portion) (S304). The voice evaluation function 104 has an influence on the sound quality when any of the following conditions is satisfied, that is, the sound quality (the sound quality of the sound reproduced by decoding the voice packet received in S301 by the receiving side codec) is poor. Evaluation (or estimation) is made (S305).

  (Condition 1) As a result of monitoring the sequence number, VoIP packets that have not arrived exceeded a certain ratio. (Condition 2) The arrival interval of VoIP packets exceeds a certain ratio of packets exceeding the size of the jitter buffer 105. (Condition 3) The sum of Condition 1 and Condition 2 exceeded a certain ratio. (Condition 4) The bit rate within a certain time is lower than a certain value compared to the bit rate of the receiving codec currently used.

As described above, it is possible to perform sound quality evaluation (or estimation).
(Details of codec selection 1)
Next, details of codec selection (S105) will be described with reference to the drawings. FIG. 5 is a flowchart for explaining details of codec selection. This is an example of a codec selection method at the time of sound quality degradation, and shows processing performed by the voice communication device 100R on the receiving side. The following processing (S400 to S404) is performed mainly by the codec selection / change request function 109, for example.

  The voice communication device 100R confirms the original bit rate and sound quality information (both provided from the codec information storage / providing function 108) of the currently used receiving side codec (S400). The voice communication device 100R extracts a codec having a bit rate lower than the confirmed bit rate of the currently used receiving side codec from the plurality of codecs 101 thereof (S401).

Next, the voice communication device 100R confirms the actual communication bit rate, and further extracts a codec having a bit rate lower than the confirmed bit rate (S402). Then, the voice communication device 100R removes the codec that has already responded incapable response (unusable answer) from the voice communication device 100T that is the other party (S403). At this point, if the codec is narrowed down to one, the voice communication device 100R selects that codec. On the other hand, if the codec is not limited to one and a plurality of codecs remain, the voice communication device 100R
Selects the one having the best sound quality from among the plurality of remaining codecs (S404).

As described above, even when the sound quality is lowered, that is, when it is evaluated (or estimated) that the sound quality has deteriorated (S104: bad), an appropriate codec is selected. (Details of codec selection 2)
Next, another example of details of codec selection (S105) will be described with reference to the drawings. FIG. 6 is a flowchart for explaining another example of details of codec selection. This is an example of a codec selection method at the time of sound quality recovery after sound quality degradation, and shows processing performed by the voice communication device 100R on the receiving side. The following processing (S500 to S505) is performed mainly by the codec selection / change request function 109, for example.

  The voice communication device 100R confirms the original bit rate and sound quality information (both provided from the codec information storage / providing function 108) of the currently used receiving side codec (S500). The voice communication device 100R extracts a codec having a bit rate higher than the confirmed bit rate of the currently used receiving side codec from the plurality of codecs 101 thereof (S501).

  Next, the voice communication device 100R confirms the actual communication bit rate, and further extracts a codec having a bit rate lower than the confirmed bit rate (S502). Then, the voice communication device 100R removes the codec having a higher bit rate than the codec used first (S503). Furthermore, the voice communication device 100R also removes the codec for which a response that cannot be handled from the voice communication device 100T that is the other party (an answer that cannot be used) has already been received (S504). At this point, if the codec is narrowed down to one, the voice communication device 100R selects that codec. On the other hand, if the number of codecs is not limited to one and a plurality of codecs remain, the voice communication device 100R selects the one with the best sound quality from the plurality of remaining codecs (S505).

As described above, an appropriate codec is selected when the sound quality is restored after the sound quality is lowered (S104: good, S106: bad).
The present invention can be implemented in various other forms without departing from the spirit or main features thereof. For this reason, said embodiment is only a mere illustration in all points, and is not interpreted limitedly.

  According to the present invention, when a call is realized by a voice packet (for example, a VoIP packet), it is assumed that the congestion state of the network (for example, an IP network) has changed over time (due to a decrease in sound quality or a subsequent recovery of sound quality). However, it is possible to realize a call with relatively good sound quality from the start to the end of the call.

It is a functional block diagram of the voice communication apparatus which is one Embodiment of this invention. It is a flowchart for demonstrating operation | movement of the audio | voice communication apparatus which is one Embodiment of this invention. It is a flowchart for demonstrating operation | movement (sound quality evaluation) of the audio | voice communication apparatus which is one Embodiment of this invention. It is a flowchart for demonstrating operation | movement (sound quality evaluation) of the audio | voice communication apparatus which is one Embodiment of this invention. It is a flowchart for demonstrating operation | movement (codec selection) of the audio | voice communication apparatus which is one Embodiment of this invention. It is a flowchart for demonstrating operation | movement (codec selection) of the audio | voice communication apparatus which is one Embodiment of this invention.

Explanation of symbols

100 voice communication device 101 codec (group)
102 Communication Function 103 Communication Status Acquisition Function 104 Sound Quality Evaluation Function 105 Jitter Buffer 106 Decoding Function 107 Audio Output Function 108 Codec Information Storage / Provision Function 109 Codec Selection / Change Request Function 110 Codec Change Request Response Function 111 Audio Input Function 112 Encoding function

Claims (6)

Receiving means for receiving voice packets including voice data encoded by the transmission side codec via a network;
Evaluation means for evaluating the sound quality of the reproduced voice by decoding the received voice packet by the receiving side codec,
Based on the evaluation result, changing means for changing the codec of each of the transmitting side and the receiving side,
A voice communication device comprising: The change means, when it is evaluated that the sound quality has deteriorated, the codec of each of the transmission side and the reception side is changed to a codec having a bit rate lower than the bit rate thereof.
The voice communication apparatus according to claim 1. When it is evaluated that the sound quality has been recovered after it is evaluated that the sound quality has deteriorated, the changing means further changes the codec after the change on each of the transmitting side and the receiving side to a bit rate higher than the bit rate of the codec. Change to rate codec,
The voice communication apparatus according to claim 2. Receiving a voice packet including voice data encoded by a transmission side codec via a network;
An evaluation step for evaluating the sound quality of the reproduced voice by decoding the received voice packet by the receiving side codec;
Based on the evaluation result, a change step of changing the codec of each of the transmission side and the reception side;
A voice communication method comprising: In the change step, when it is evaluated that the sound quality is deteriorated, the codec of each of the transmission side and the reception side is changed to a codec having a lower bit rate than the bit rate thereof.
The voice communication method according to claim 4. In the changing step, when it is evaluated that the sound quality is recovered after it is evaluated that the sound quality is deteriorated, the codec after the change on each of the transmitting side and the receiving side is further changed to a bit higher than the bit rate of the codec. Change to rate codec,
The voice communication method according to claim 5.

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