JP2015015741A - Media communication system, media communication device, media communication method, and media communication program - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a media communication system capable of preventing deterioration in communication quality of media data even if the media data has gone through a network causing transmission delay at a period right after starting transmission and reception of the media data.SOLUTION: A media communication system of the invention comprises a plurality of media communication devices that exchange call control signals by a procedure in accordance with a call control protocol, and thereby establish a session with opposite media communication devices to exchange media data. Each of the plurality of media communication devices comprises: dummy data transmission means that transmits dummy media data before establishment of a session, whether it is a call originating side or a call incoming side of an initial call control signal for session establishment; and original data transmitting/receiving means for transmitting and receiving original media data after the establishment of the session.

Description

  The present invention relates to a media communication system, a media communication apparatus, a media communication method, and a media communication program, and is applicable 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 system, a media communication method, and a media communication program that can prevent the deterioration of the communication quality of media data even during a period immediately after the start of transmission / reception of media data through a network that causes transmission delay are desired. It is rare.

  The media communication system according to the first aspect of the present invention provides a plurality of media communications for establishing a session with an opposing media communication device and exchanging media data by exchanging a call control signal according to a procedure according to a call control protocol. In the media communication system provided with the apparatus, each of the plurality of media communication apparatuses is (1) whether the originating side or the terminating side of the original call control signal for session establishment, before establishing the session. And dummy data transmitting means for transmitting dummy media data, and (2) original data transmitting / receiving means for transmitting and receiving original media data after the session is established.

  A media communication apparatus according to a second aspect of the present invention provides 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. (1) Dummy data transmission for transmitting dummy media data before the session is established, regardless of whether the media communication apparatus is the original call control signal originator or receiver of the session establishment. And (2) original data transmission / reception means for transmitting / receiving original media data after the session is established, and (3) if the media communication device is a call control signal transmission side, the dummy data The transmission means transmits dummy media data before the INVITE request. (4) The media communication apparatus receives the call control signal. If it is, the dummy data transmitting unit, and transmits a dummy media data before Ringing response.

  A media communication method according to a third aspect of the present invention establishes a session with an opposing media communication device by exchanging a call control signal between a plurality of media communication devices in a procedure according to a call control protocol. In the media communication method for exchanging data, (1) the dummy data transmission means receives the incoming call even if each media communication apparatus equipped with the dummy data transmission means is the originator of the original call control signal for session establishment. Even on the side, dummy media data is transmitted before the session is established, and (2) the original data transmitting / receiving means transmits and receives the original media data after the session is established.

  A media communication program according to a fourth aspect of the present invention establishes a session with an opposing media communication device by exchanging a call control signal between a plurality of media communication devices in a procedure according to a call control protocol. (1) Establishing a session regardless of whether the media communication device is the originator or receiver of the original call control signal for establishing a session. It is characterized by functioning as dummy data transmission means for transmitting dummy media data before and (2) original data transmission / reception means for transmitting and receiving original media data after the session is established.

  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 system, a media communication apparatus, a media communication method, and a media communication program according to the present invention 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 point of the dummy RTP packet from the IP telephone 1R side is immediately before the transmission point of the 180 Ringing response, but even after the transmission of the 180 Ringing response and before the transmission of the 200 OK 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 is apparent from the above, in both the originating IP telephone 1S and the terminating IP telephone 1R, the transmission time of the normal RTP packet related to the call starts almost continuously with the transmission period of the dummy RTP packet.

(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 referred to. However, modified embodiments as exemplified below can be cited.

  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 (5)

In a media communication system having a plurality of media communication devices that exchanges media data by establishing a session with an opposing media communication device by exchanging a call control signal in a procedure according to a call control protocol.
Each of the plurality of media communication devices
Dummy data transmission means for transmitting dummy media data before the establishment of the session, whether it is the originating side or the terminating side of the initial call control signal for session establishment,
A media communication system comprising: original data transmission / reception means for performing transmission / reception of original media data after establishment of the session.   When the media communication device is the call control signal originator, the dummy data transmission means transmits dummy media data before the INVITE request, and the media communication device is the call control signal destination. 2. The media communication system according to claim 1, wherein the dummy data transmitting means transmits dummy media data before a Ringing response. 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.
Dummy data transmission means for transmitting dummy media data before the establishment of the session, whether the media communication device is a caller side or a callee side of an initial call control signal for session establishment;
After the establishment of the session, it is provided with original data transmission / reception means for transmitting / receiving original media data,
If the media communication device is the call control signal originator, the dummy data sending means sends dummy media data before the INVITE request,
When the media communication device is a call control signal receiving side, the dummy data transmission means transmits dummy media data before the Ringing response. 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 between a plurality of media communication devices.
The dummy data transmission means is provided before the establishment of the session, regardless of whether each media communication device equipped with the dummy data transmission means is the original call control signal originator or receiver of the session establishment. , Send dummy media data,
The original data transmission / reception means transmits / receives original media data after the session is established. Installed in each media communication device that establishes a session with the opposite media communication device and exchanges media data by exchanging a call control signal between a plurality of media communication devices in a procedure according to a call control protocol. Computer
Dummy data transmission means for transmitting dummy media data before the establishment of the session, whether the media communication device is a caller side or a callee side of an initial call control signal for session establishment;
A media communication program which functions as an original data transmitting / receiving means for transmitting / receiving original media data after the session is established.

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