JP5696514B2 - Media communication apparatus, method and program, and media communication system - Google Patents

Description

  The present invention relates to a media communication apparatus, method and program, and a media communication system, and can be applied to, for example, P2P voice communication between IP telephones (including softphones) using VoIP technology.

  In recent years, IP telephone communication, which is voice communication using VoIP technology, has become widespread. For connection between IP telephones, the SIP protocol is often used. For example, a procedure as shown in Non-Patent Document 1 is performed. This procedure is described below.

  When the caller lifts the handset and dials (calls) the telephone number of the callee, an INVITE request is sent to the callee. On the called party side, a ringing bell rings and a 180 Ringing response indicating that the calling is in progress is sent to the calling party. On the caller side, a “ringing sound” is played by the 180 Ringing response, and it is notified that the callee is being called. When the callee picks up the handset and responds, a 200 OK response is sent to the caller. By sending an ACK request from the caller to the callee, a media session (line connection) for communication is established by RTP (Real Time Data Transfer Protocol), and a call can be made.

Supervised by Yasuaki Chimura and Toshifumi Murata, “Revised SIP Textbook”, published by Impress, Inc., December 2004, p. 82

  As described above, actual voice communication (transmission / reception of RTP packet) is started after transmission of the ACK request.

  However, depending on the network to which RTP packets are transmitted and received, as shown in FIG. 3, even if the transmitting side continuously transmits RTP packets for a while after the start of voice communication (call), Each RTP packet does not arrive in real time, and a plurality of RTP packets transmitted immediately after the start of voice communication (call) arrive at the receiving side collectively. In such a network, a call delay occurs immediately after the start of a call, resulting in a deterioration in call quality.

  Therefore, a media communication apparatus, method and program, and a media communication system capable of preventing deterioration in communication quality of media data even during a period immediately after the start of transmission / reception of media data via a network causing transmission delay are desired. It is rare.

According to a first aspect of the present invention, there is provided a media communication apparatus that establishes a session with an opposite media communication apparatus and exchanges media data by exchanging a call control signal according to a procedure according to a call control protocol. (2) Dummy data transmission means for transmitting dummy media data before establishment of (2) Original data transmission / reception means for transmitting / receiving original media data after establishment of the session, and (3) Opposing communication partner (4) The dummy data transmission unit uses the specific information stored in the storage unit for transmission of the dummy media data. It is characterized by that.

The second of the present invention, by exchanging call control signals with the procedure in accordance with the call control protocol, in the media communication method for exchanging media data to establish a facing media communication apparatus and the session, (1) memory The means stores in advance the identification information of the opposing media communication device as the communication partner, and (2) the dummy data transmission means uses the identification information stored in the storage means to establish the session. Before the transmission of dummy media data, (3) the original data transmission / reception means transmits / receives the original media data after the session is established.

According to a third aspect of the present invention, there is provided a computer mounted on a media communication apparatus that establishes a session with an opposite media communication apparatus and transmits / receives media data by exchanging a call control signal according to a procedure according to a call control protocol. (1) Storage means that stores in advance the specific information of the opposing media communication device that is the communication partner, and (2) using the specific information stored in the storage means, before establishing the session, It is characterized by functioning as dummy data transmission means for transmitting dummy media data, and (3) original data transmission / reception means for transmitting and receiving original media data after the establishment of the session.

The media communication system of the fourth aspect of the present invention is characterized in that the two media communication apparatuses of the first aspect of the present invention face each other and exchange media data.

  According to the present invention, transmission / reception of dummy media data is started prior to transmission / reception of original media data, thereby preventing a transmission delay from occurring immediately after the start of transmission of original media data. be able to.

It is a block diagram which shows the functional structure of the IP telephone of embodiment. FIG. 3 is a sequence diagram showing a procedure for exchanging signals between a calling IP phone and a called IP phone in the voice communication system of the embodiment. It is explanatory drawing of the initial delay of the network through which RTP packet is transmitted / received.

(A) Main Embodiment Hereinafter, an embodiment of a media communication apparatus, method and program according to the present invention and a media communication system will be described with reference to the drawings. The media communication device of this embodiment is an IP telephone (including a soft phone) using VoIP technology.

(A-1) Configuration of Embodiment FIG. 1 is a block diagram illustrating a functional configuration of an IP telephone according to an embodiment. Even when the IP telephone is composed of a soft phone (comprising a CPU and a program executed by the CPU), it can be functionally represented in FIG.

  Two IP telephones 1 having the configuration shown in FIG. 1 face each other via an IP network N, and a voice communication system is formed to perform voice communication (call).

  In FIG. 1, an IP telephone 1 according to an embodiment includes a call control unit 2, a microphone 3, a voice encoding unit 4, an RTP packet transmission unit 5, an RTP packet reception unit 6, a voice decoding unit 7, and a speaker. 8.

  The call control unit 2 executes call control such as establishment of a media session (line connection) for communication according to the SIP protocol. The call control unit 2 of this embodiment includes a communication partner IP address storage unit 10 and a dummy RTP packet period control unit 11 in addition to a general configuration.

  The communication partner IP address storage unit 10 stores an IP address that can be a communication partner of P2P communication. Note that when a call operation is performed, an IP address of a communication partner may be fetched from another device before starting the SIP protocol procedure.

  The dummy RTP packet period control unit 11 instructs the RTP packet transmission unit 5 to start or end transmission of the dummy RTP packet. The timing for instructing the start of transmission and the end of transmission will be clarified in the description of operations described later.

  The microphone 3 captures the voice of the near-end speaker, and the speech encoding unit 4 converts the speech signal obtained by the microphone 3 into a digital signal and then encodes it. As the microphone 3 and the voice encoding unit 4, existing ones can be applied as they are. Note that the timing at which the speech encoding unit 4 starts processing and the timing at which processing ends is controlled by the call control unit 2.

  The RTP packet transmission unit 5 basically forms and transmits an RTP packet in which the encoded voice data output from the voice encoding unit 4 is inserted. The RTP packet transmission unit 5 includes an RTP packet assembly unit 20, a dummy RTP packet output unit 21, and an output RTP packet switching unit 22.

  The RTP packet assembling unit 20 assembles an RTP packet into which the encoded audio data output from the audio encoding unit 4 is inserted. The IP address given from the call control unit 2 is inserted as the counterpart address in the RTP packet.

  The dummy RTP packet output unit 21 outputs a dummy RTP packet including dummy data without including the encoded voice data formed from the voice signal of the near-end speaker. The IP address given from the call control unit 2 is inserted as the counterpart address in the dummy RTP packet. As will be described later, the dummy RTP packet is assumed to be discarded by the opposite IP telephone. However, for example, background data may be used as dummy data so that sound may be output without being discarded. Data encoded with noise can be applied. The dummy RTP packet can determine that the RTP packet is a dummy RTP packet by using a part of the data part or a part of the header part.

  Under the control of the call control unit 2, the output RTP packet switching unit 22 transfers one of the dummy RTP packet output from the dummy RTP packet output unit 21 or the dummy RTP packet output from the RTP packet assembly unit 20 to the IP network N. To send to.

  The RTP packet receiving unit 6 basically extracts the encoded voice data inserted in the RTP packet that has arrived from the IP network N, and gives it to the voice decoding unit 7. The RTP packet receiving unit 6 includes an RTP packet determining unit 30 and an RTP packet decomposing unit 31.

  The RTP packet discriminating unit 30 discriminates whether the RTP packet arriving from the IP network N is a dummy RTP packet or another normal RTP packet. If it is a dummy RTP packet, it is discarded, and if it is a normal RTP packet, This is given to the RTP packet decomposition unit 31.

  The RTP packet decomposing unit 31 extracts the encoded audio data inserted in the RTP packet given from the RTP packet determining unit 30 and gives it to the audio decoding unit 7.

  The audio decoding unit 7 decodes the encoded audio data given from the RTP packet decomposing unit 31, further converts it into an analog signal (audio signal) and gives it to the speaker 8, and the speaker 8 Sound output. As the audio decoding unit 7 and the speaker 8, existing ones can be applied as they are.

(A-2) Operation of Embodiment Next, the operation of the audio communication system (media communication system) of the embodiment will be described with reference to the drawings. In other words, the communication operation between the IP telephones of the two embodiments will be described. In the following, in order to distinguish between two IP telephones, “S” is further added to the reference numeral shown in FIG. 1 for the originating IP telephone and its constituent elements, and the incoming IP telephone and its constituent elements are given. On the other hand, “R” is further given to the reference numerals shown in FIG.

  FIG. 2 is a sequence diagram showing a procedure for exchanging signals between the IP telephone 1S on the outgoing side and the IP telephone 1R on the incoming side. FIG. 2 also shows a case in accordance with the SIP protocol.

  When the call control unit 2S of the IP telephone 1S recognizes that the user (near-end speaker) has made a call operation to the IP telephone 1R, first, the IP address of the IP telephone 1R is read from the communication partner IP address storage unit 10S. The dummy RTP packet period control unit 11S instructs the dummy RTP packet output unit 21S and the output RTP packet switching unit 22S to transmit the dummy RTP packet while giving the extracted IP address (step 100). From this time, the dummy RTP packet output unit 21S outputs the dummy RTP packet, and the output RTP packet switching unit 22S starts selecting and transmitting the dummy RTP packet to the IP network N.

  When the call control unit 2S of the IP telephone 1S recognizes that the user (near-end speaker) has made a call operation to the IP telephone 1R, the call control unit 2S uses the IP address extracted from the communication partner IP address storage unit 10S. An INVITE request including the destination address is transmitted to the IP network N (step 101).

  Although FIG. 2 shows the case where the transmission start time of the dummy RTP packet from the IP telephone 1S side is immediately before the transmission time of the INVITE request, any of these contexts may be first. In short, the transmission of the dummy RTP packet may be started almost in parallel with the processing operation at the time of transmission.

  When the INVITE request arrives, the call control unit 2R of the receiving side IP telephone 1R determines whether or not the response can be made. If the response is possible, the call control unit 2R starts transmission of the dummy RTP packet (step 102). The ringing bell is ringed, and a 180 Ringing response indicating that the call is being made is transmitted to the caller (step 103). Also in the IP telephone 1R on the receiving side, when the transmission of the dummy RTP packet is started, the dummy RTP packet output unit 21R outputs the dummy RTP packet, and the output RTP packet switching unit 22R selects the dummy RTP packet and transmits it to the IP network N. Start sending. In this case, the destination address of the dummy RTP packet is the IP address of the transmission source of the INVITE request.

  The call control unit 2R of the incoming side IP telephone 1R transmits a 200 OK response to the outgoing side IP telephone 1S when the receiver raises the handset and responds (step 104).

  FIG. 2 shows the case where the transmission start time point of the dummy RTP packet from the IP telephone 1R side is immediately before the transmission time point of the 180 Ringing response. Alternatively, it may be immediately after the transmission of the 200 OK response. In short, it suffices to start the transmission of the dummy RTP packet substantially in parallel with the initial process at the time of the incoming call.

  Unlike FIG. 2, if the IP telephone 1 </ b> R cannot respond due to busy or the like, transmission of the dummy RTP packet is not started and transmission is immediately terminated if transmission of the dummy RTP packet is started. At the same time, a response indicating that the response cannot be made (the content of the response varies depending on the reason) is transmitted to the IP telephone 1S on the transmission side.

  Although different from FIG. 2, when the call control unit 2S of the originating IP telephone 1S receives a response indicating that it cannot respond from the incoming IP telephone 1R, the dummy RTP packet output unit 21S and the output RTP packet switching unit 22S To end transmission of the dummy RTP packet.

  On the other hand, when receiving a 200 OK response from the incoming IP telephone 1R, the call control unit 2S of the outgoing IP telephone 1S transmits an ACK request to the incoming IP telephone 1R (step 105), and RTP packet The transmission unit 5S terminates the transmission of the dummy RTP packet, instructs transmission of the RTP packet related to the call, and shifts to the call state (step 106).

  When receiving the ACK request, the call control unit 2R of the IP telephone 1R on the receiving side terminates the transmission of the dummy RTP packet to the RTP packet transmission unit 5R, instructs the transmission of the RTP packet related to the call, (Step 106).

  In the originating IP telephone 1S, the dummy RTP packet that arrives before entering the call state is determined to be a dummy RTP packet by the RTP packet determining unit 30S and discarded. Similarly, in the IP telephone 1R on the receiving side, the dummy RTP packet that has arrived before entering the call state is determined to be a dummy RTP packet by the RTP packet determining unit 30R and discarded.

  Note that the call control unit 2S or 2R instructs the RTP packet reception unit 6S or 6R to discard the RTP packet that arrives before entering the call state without determining that it is a dummy RTP packet. The RTP packet may be discarded.

  When the IP telephone 1S on the transmission side shifts to a call state, the voice signal obtained by the microphone 3S is encoded after being converted into a digital signal by the voice encoding unit 4S, and the RTP packet assembling unit 20S is obtained. An RTP packet in which encoded voice data is inserted is formed, and this RTP packet is transmitted to the destination IP telephone 1R via the output RTP packet switching unit 22S. Similarly, when the incoming side IP telephone 1R shifts to a call state, the voice signal obtained by the microphone 3R is encoded after being converted into a digital signal by the voice encoding unit 4R, and the RTP packet assembling unit 20R is encoded. The RTP packet into which the encoded voice data obtained is inserted is formed, and this RTP packet is transmitted to the IP telephone 1S on the transmission side via the output RTP packet switching unit 22R.

As apparent from the above, also in the IP telephone 1S calling party, even in the IP telephone 1R incoming side, substantially continuously to transmit period of the dummy RTP packet, the transmission time of a normal RTP packet representing the call Begins.

(A-3) Effect of Embodiment According to the above embodiment, since transmission / reception of a dummy RTP packet is executed prior to transmission / reception of an RTP packet by an actual call, a large communication delay occurs when the network starts transmission / reception of the RTP packet. Even in the call period, even during the call period, it is not affected by the characteristics of the network, and good call quality can be achieved throughout the call period.

(B) Other Embodiments In the description of the above-described embodiment, various modified embodiments have been mentioned.

  In the above embodiment, the transmission of a dummy RTP packet prepared in advance is shown before the call period. However, the RTP including information of the audio signal obtained by the microphone before the call period is also shown. A packet may be transmitted. Even in this case, it is necessary to discard the received RTP packet before the call period. The term “dummy” in the claims includes such a case.

  In the above embodiment, the IP phone on the receiving side uses the reception condition of the INVITE request as a transmission condition of the dummy RTP packet. However, when the dummy RTP packet transmitted by the IP phone on the calling side starts to arrive, the INVITE Even before the request is received, transmission of the dummy RTP packet may be started.

  In the above embodiment, the destination of the dummy RTP packet is the IP telephone of the communication partner, but another apparatus may be used.

  In the above embodiment, the media communication apparatus is an IP telephone, but any apparatus that exchanges media data using RTP packets may be used. That is, the media data is not limited to an audio signal, and may be an image signal or a data signal.

  DESCRIPTION OF SYMBOLS 1 ... IP telephone, 2 ... Call control part, 5 ... RTP packet transmission part, 6 ... RTP packet receiving part, 10 ... Communication partner IP address storage part, 11 ... Dummy RTP packet period control part, 20 ... RTP packet assembly part, 21 ... Dummy RTP packet output unit, 22 ... Output RTP packet switching unit, 30 ... RTP packet determination unit, 31 ... RTP packet decomposition unit.

Claims (9)

In a media communication apparatus that establishes a session with an opposing media communication apparatus and exchanges media data by exchanging a call control signal in a procedure according to a call control protocol.
Before establishing the session, dummy data transmission means for transmitting dummy media data;
After the session is established, original data transmission / reception means for transmitting / receiving original media data;
Comprising storage means for storing in advance the specific information of the opposing media communication device as the communication partner;
The dummy data transmitting means uses the specific information stored in the storage means for transmitting the dummy media data.   2. The media communication apparatus according to claim 1, further comprising dummy data discarding means for discarding dummy media data that arrives before the session is established.   The media communication apparatus according to claim 1 or 2, wherein the dummy media data and the original media data are transmitted and received by an RTP packet. In a media communication method for establishing a session with an opposing media communication device and exchanging media data by exchanging a call control signal in a procedure according to a call control protocol,
The storage means stores in advance the specific information of the opposing media communication device that is the communication partner,
The dummy data transmission means uses the specific information stored in the storage means to transmit dummy media data before the establishment of the session,
The original data transmission / reception means transmits / receives original media data after the session is established. Dummy data discarding means, media communication method according to claim 4, wherein the benzalkonium discard the dummy media data arriving before the establishment of the session.   6. The media communication method according to claim 4, wherein the dummy media data and the original media data are transmitted and received by an RTP packet. A computer mounted on a media communication apparatus that establishes a session with an opposing media communication apparatus and exchanges media data by exchanging a call control signal in a procedure according to a call control protocol,
Storage means for preliminarily storing specific information of an opposing media communication device as a communication partner;
Using the specific information stored in the storage means, before establishing the session, send dummy media data, dummy data transmission means,
A media communication program which functions as an original data transmission / reception means for transmitting / receiving original media data after the session is established. 8. The media communication program according to claim 7, wherein the computer functions as dummy data discarding means for discarding dummy media data that arrives before the session is established. 2. A media communication system, wherein the two media communication devices according to claim 1 face each other and exchange media data.

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