JP2009260719A - Data transmission terminal device and data transmission method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a data transmission terminal device for transmitting, in real time, audio or video to another terminal through a wireless network without increasing a load on a CPU or the network. <P>SOLUTION: A terminal 11 transmits an audio realtime transport protocol (RTP) packet containing audio data of 160 bytes at intervals of 20 ms to the other terminal through a wireless network. If a transmission error of the packet sent to the wireless network occurs at a certain time, the terminal 11 does not retransmit the same audio RTP packet immediately but holds the retransmission until the next packet transmission timing comes. When the next packet transmission timing comes, the terminal 11 couples audio data in the preceding audio RTP packet and audio data to be next transmitted, thereby transmitting one audio RTP packet containing audio data of 320 bytes (=160+160) to the other terminal. A re-transmissible maximum packet size is limited within a jitter buffer size of the other terminal. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to a data transmission terminal apparatus and a data transmission method for transmitting audio or video in real time.

  In recent years, VoIP (Voice over IP) technology for transmitting voice data in real time using an IP network such as the Internet has begun to be used. In this VoIP technology, a real-time transfer protocol (RTP) and UDP are used. With this VoIP technology, an IP telephone system can be realized.

  In the IP telephone system, not only audio data but also video data can be transmitted. By transmitting audio data and video data, the IP telephone system can also function as a TV telephone system.

  In a normal IP telephone system, it is assumed that the terminal is connected to a wired LAN. Recently, in order to enhance mobility, it is desired to realize a technology (wireless LAN IP phone) for performing a voice call via a wireless network such as a wireless LAN.

  The wireless LAN IP phone is convenient in terms of mobility, but the wireless LAN communication part is unstable compared to wired LAN communication, and the communication quality is considerably inferior. For this reason, when implementing a wireless LAN IP telephone, a mechanism for improving the reliability of packet transmission is required.

Patent Document 1 discloses a data communication apparatus that retransmits data of a packet that has failed to be transmitted when an acknowledgment cannot be received from a communication partner within a predetermined time. In this data communication apparatus, when a packet transmission error occurs, the packet is immediately retransmitted. If a transmission error occurs repeatedly and the data of the packet to be transmitted next is prepared while the retransmission of the packet is repeated many times, the data of the packet that failed to be transmitted and the next transmission The data of the packet to be reconstructed is reconstructed as one packet and transmitted. The maximum size of the reconstructed packet is defined with the maximum frame size (for example, 1500 bytes) from the wireless access point as the upper limit.
JP 2007-116411 A

  However, in the case of using a wireless network, if the retransmission control of immediately resending the same packet data as in Patent Document 1 is used, a lot of load may be wasted on the CPU and network of the transmitting terminal. .

  The reason is that the occurrence of a transmission error means that the wireless environment is very congested, and if it is retransmitted immediately, there is a risk that the congestion will be spurred. In addition, when there is a failure between the terminal and the router in the wireless network, even if it is immediately retransmitted, the communication environment has not changed significantly in a short time, and it is expected that there is a low probability that it can be relieved.

  In Patent Document 1, the maximum size of the reconstructed packet is defined as the maximum frame size with the wireless access point regardless of the communication environment of the counterpart terminal. Causes a problem such as buffer overflow, so that the voice data in the reconstructed packet cannot be reproduced in real time on the receiving side.

  The present invention has been made in consideration of the above-mentioned circumstances, and is a data transmission terminal device capable of transmitting audio and video to a partner terminal in real time via a wireless network without increasing the load on the CPU or network. And it aims at providing a data transmission method.

  In order to solve the above-described problem, the present invention provides a data transmission terminal device for transmitting audio or video in real time to a counterpart terminal via a wireless network, the packet having a predetermined size including the audio or video data Packet transmission means for transmitting to the counterpart terminal via the wireless network at a predetermined time interval, and when a transmission error occurs for a packet sent to the wireless network, until the next packet transmission timing arrives Holds retransmission of the packet in which the transmission error has occurred and generates one packet by combining the data in the packet immediately before the transmission error and the next data to be transmitted at the next packet transmission timing And re-transmitting the generated packet to the partner terminal via the wireless network. Data transmission terminal apparatus characterized by comprising a control means.

  The present invention is also a data transmission terminal device for transmitting audio or video in real time to a partner terminal via a wireless network, wherein the packet of a predetermined size including the audio or video data is transmitted to the wireless network. The packet transmission means for transmitting to the counterpart terminal via a predetermined time interval and the acknowledgment for the packet sent to the wireless network is not returned from the router in the wireless network, the acknowledgment is returned Means for generating a retransmission count over event indicating that a transmission error has occurred for the packet when the packet retransmission count exceeds a predetermined value, and the retransmission count over event occurs. The number of retransmissions until the next packet transmission timing arrives By holding the retransmission of the packet in which the transmission error has occurred by the server, and combining the data in the packet immediately before the transmission error due to the number of retransmissions and the next data to be transmitted at the next packet transmission timing Retransmission control means for generating one packet and transmitting the generated packet to the counterpart terminal via the wireless network, and determining the buffer size for jitter absorption of the counterpart terminal by communication with the counterpart terminal And a maximum packet size generated by the retransmission control unit is limited to be equal to or smaller than the determined buffer size.

  According to the present invention, audio and video can be transmitted to a counterpart terminal in real time via a wireless network without increasing the load on the CPU or network.

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

  FIG. 1 shows a configuration example of a data transmission system using a data transmission terminal device according to an embodiment of the present invention. This data transmission system is a system (wireless LAN IP telephone system) that transmits data that requires real-time transmission such as voice and video via a wireless network such as a wireless LAN.

  In this wireless LAN IP telephone system, a terminal (A) 11 is connected to a wireless network such as a wireless LAN, and passes through a wireless router (also referred to as a wireless LAN router or an access point) 12. Communication with the Internet (IP network) 13 is executed. Similarly, the terminal (B) 15 is also connected to a wireless network such as a wireless LAN, and performs communication with the Internet (IP network) 13 via the wireless router 14.

  The terminal (A) 11 functions as a data transmission terminal device of this embodiment. The terminal (A) 11 is composed of an information processing terminal such as a personal computer, for example. The terminal (A) 11 functions as an IP telephone terminal, and is configured to execute audio or video transmission / reception in real time with a counterpart terminal (for example, the terminal (B) 15) via a wireless network.

  In the present embodiment, transmission of data such as voice or video between the terminal (A) 11 and the terminal (B) 12 is performed using a real-time transfer protocol (RTP) or an RTP control protocol (RTCP: RTP Control). Protocol), UDP / IP protocol, etc.

  In general, a wired LAN IP phone is a G. 711 is used to transmit and receive audio data using RTP (RFC3550). G. If the sampling frequency is 8000 Hz and the voice packet transmission interval is 20 ms, 711 is as follows.

・ Sampling frequency: 8000Hz
-Principle delay: 0.125 ms x number of jitter buffers-Number of packets sent per second: 1000/20 ms = 50 pps
-Required bandwidth per audio channel 64 Kbps = 50 pps x Payload size-Payload size = 64000/50 = 1280 bits = 160 bytes
Voice packet length = about 200 bytes RTCP (RFC3550) is used to control the flow of voice RTP packets and provides control information. The most important function of RTCP is to provide QoS feedback provided by RTP.

  As described above, the wireless LAN IP telephone system is convenient in terms of mobility, but the wireless LAN communication part is unstable compared to wired LAN communication, and the communication quality is considerably inferior.

  The terminal (A) 11 transmits voice and video data to the partner terminal (B) 12 in real time via the wireless LAN without imposing much load on the CPU and network in the terminal (A) 11. It has a function to make it possible.

  Hereinafter, the data transmission operation by the terminal (A) 11 will be described by exemplifying a case where audio data is transmitted from the terminal (A) 11 to the terminal (B) 15 in real time.

  RTP packets are used for transmission of voice packets. The payload of the voice RTP packet includes, for example, 160 bytes of voice data. The transmission interval of the voice RTP packet is 20 ms, for example. That is, voice data for 20 ms is transmitted to the other terminal by one voice RTP packet. On the other hand, RTCP packet transmission for providing QoS feedback between the transmitting terminal and the partner terminal is executed at a rate of about once every 2 s, for example.

  The terminal (A) 11 transmits a packet including voice data of 160 bytes (voice RTP packet) to the other terminal via the wireless network at intervals of 20 ms. When a transmission error occurs for a packet sent to the wireless network at a certain time, the terminal (A) 11 does not immediately retransmit the same voice RTP packet, but a transmission error occurs until the next packet transmission timing arrives. The retransmission of the received voice RTP packet is suspended. This is because even if it is immediately retransmitted, the communication environment of the wireless network is not always improved in a short time, and there is a high possibility that a transmission error will occur again. In this case, useless packet retransmission is performed, resulting in an increase in the load on the CPU and the network. In FIG. 1, a voice RTP packet in which a transmission error has occurred is indicated by a cross.

  When the next packet transmission timing arrives, the terminal (A) 11 retransmits the packet in which the previous transmission error has occurred. In this case, the terminal (A) 11 combines the voice data (160 bytes) in the voice RTP packet immediately before the transmission error has occurred with the voice data (160 bytes) to be transmitted next, thereby 320 bytes ( = 160 + 160) is generated, and the generated voice RTP packet is transmitted to the counterpart terminal via the wireless network.

  If the transmission of the voice RTP packet including the 320-byte voice data is successful, a normal voice RTP packet including only the 160-byte voice data to be transmitted next is transmitted at the next packet transmission timing. On the other hand, when transmission of a packet containing 320-byte audio data fails, the terminal (A) 11 again holds retransmission until the next packet transmission timing arrives. When the next packet transmission timing arrives, the terminal (A) 11 combines the data (160 + 160 bytes) in the packet immediately before the transmission error occurs with the voice data (160 bytes) to be transmitted next. Thus, one voice RTP packet including voice data of 480 bytes (= 160 + 160 + 160) is generated, and the generated voice RTP packet is transmitted to the counterpart terminal via the wireless network.

  If the transmission of the voice RTP packet including the 480-byte voice data is successful, a normal voice RTP packet including only the 160-byte voice data to be transmitted next is transmitted at the next packet transmission timing. On the other hand, when the transmission of the voice RTP packet including the voice data of 480 bytes fails, the terminal (A) 11 again holds the retransmission until the next packet transmission timing arrives. When the next packet transmission timing arrives, the terminal (A) 11 combines the data (160 + 160 + 160 bytes) in the voice RTP packet immediately before the transmission error occurs and the voice data (160 bytes) to be transmitted next. As a result, one voice RTP packet including 640 bytes (= 160 + 160 + 160 + 160) of voice data is generated, and the generated voice RTP packet is transmitted to the other terminal via the wireless network.

  As described above, in the retransmission control according to the present embodiment, the retransmission of the immediately preceding packet is suspended until the next packet transmission timing, so that it is possible to prevent the retransmission of the packet from being repeated many times. In the present embodiment, although the timing at which a packet whose transmission has failed reaches the counterpart terminal increases in units of 20 ms, the counterpart terminal is usually provided with a buffer for absorbing jitter (jitter buffer). Delays in packet arrival timing can be absorbed.

  In the present embodiment, the maximum packet size (maximum number of packets to be combined) generated when packet transmission errors continuously occur at a plurality of continuous packet transmission timings is equal to or smaller than the buffer size of the jitter buffer of the partner terminal. Limited. For example, when the jitter buffer size of the partner terminal is 640 bytes (= 4 packets), the maximum packet size (maximum number of packets to be combined) is limited to 640 bytes (= 4 packets). In this way, by defining the upper limit of the maximum packet size (maximum number of packets to be combined) that can be generated in consideration of the size of the jitter buffer of the counterpart terminal, it is possible to guarantee real-time transmission.

  FIG. 2 shows a functional configuration of the terminal (A) 11.

  The terminal (A) 11 includes a voice input unit 111, an IP telephone control unit 112, an RTP transmission processing unit 113, an IP transmission processing unit 114, a wireless communication unit 115, and the like. The voice input unit 111 inputs user voice data. The voice input unit 111 samples user's voice input from a microphone, for example, and converts it into voice PCM data.

  The IP telephone control unit 112 controls voice transmission / reception using a real-time transfer protocol or the like. The IP telephone control unit 112 performs voice transmission / reception and retransmission control at the application layer level. The IP telephone control unit 112 encodes audio data input from the audio input unit 111 every 20 ms, and converts the encoded audio data packet into an RTP transmission processing unit 113, an IP transmission processing unit 114, and a wireless communication unit. 115. This IP telephone control unit 112 is provided with the above-described retransmission control function and a maximum packet size limiting function for limiting the maximum packet size in consideration of the jitter buffer size of the counterpart terminal.

  The RTP transmission processing unit 113 adds an RTP header to the packet generated by the IP telephone control unit 112 and generates an RTP packet. The RTP header can include information such as a packet transmission time and a sequential number. The IP transmission processing unit 114 generates a UDP packet by adding a UDP header to the RTP packet. The UDP header can include the IP address of the partner terminal. The UDP packet is sent to the wireless communication unit 115 via a wireless LAN driver that controls the wireless communication unit 115. The wireless communication unit 115 is a wireless communication device for executing wireless communication, and functions as a physical layer (MAC layer). The wireless communication unit 115 transmits and receives packets to and from the wireless router 12 by wireless signals.

  In the present embodiment, if the wireless communication unit (MAC layer) 115 does not return an acknowledgment (ACK response) to a packet sent to the wireless network at a certain time from the wireless router 12, it returns until an acknowledgment is returned. It is configured to repeat retransmission of packets. When the number of packet retransmissions exceeds a predetermined number, wireless communication unit (MAC layer) 115 generates a retransmission number over event and notifies IP telephone control unit 112 that a packet transmission error has occurred. . That is, the wireless communication unit (MAC layer) 115 confirms transmission arrival with an acknowledgment (ACK response) from the wireless router 12 for the data transmission packet of the MAC layer. If there is no ACK response, the wireless communication unit (MAC layer) 115 retransmits the same data packet, and if it cannot receive an ACK response within the specified number of times, it detects an abnormality as a transmission retransmission over in the MAC layer. By such a retransmission mechanism at the MAC layer level, lost packets on the air can be considerably saved without increasing the load on the CPU in the terminal 11.

  The IP phone control unit 112 functioning as an application layer receives a packet over-reception event from the wireless communication unit (MAC layer) 115 on the condition that the number of packet retransmissions exceeds a predetermined value. It is determined that a transmission error has occurred, and the process proceeds to retransmission control processing at the application layer level.

  FIG. 3 shows a transmission flow when retransmission succeeds once.

  The voice PCM data input from the microphone is encoded by the IP telephone control unit 112, and a packet containing 160-byte voice data is generated. This packet is sent to the wireless network after the RTP header, UDP header, LLC header, and MAC header are added. If the retransmission succeeds once, an ACK response is returned from the wireless router 12 to the wireless communication unit 115 within a predetermined time from the first retransmission.

  In the present embodiment, the IP telephone control unit 112 can retransmit the same voice data immediately when the retransmission is over in the MAC layer, but does not perform such processing as described above. That is, the occurrence of retransmission over at the MAC layer is because the wireless environment is very congested, and retransmission at the application layer level may add to the congestion. In addition, when there is a failure between the terminal (A) 11 and the wireless router 12, it is expected that there is a low probability that the communication environment will not change significantly in a short time even if it is retransmitted in units of several tens of milliseconds. Because it is done. If retransmission at the application layer level by the IP telephone control unit 112 is performed in a poor communication environment, retransmission at the MAC layer level is repeated again and again, which increases the load on the wireless network. become.

  FIG. 4 shows retransmission overflow when the number of retransmissions is five.

  In this embodiment, when retransmission over occurs at the MAC layer level, the upper layer IP telephone control unit 112 performs the following retransmission control.

  1. When a voice packet re-transmission over occurs, the IP telephone control unit 112 determines that the packet to be transmitted next is the previous re-transmission over packet at the next voice RTP packet transmission timing (for example, 20 ms later in G.711). Combined and sent as a packet together.

  2. When retransmission over occurs continuously, it is combined with a plurality of retransmission over packets so far and transmitted as a packet together.

  3. The maximum number of packets to be combined is determined based on the jitter buffer size of the partner terminal.

  FIG. 5 shows a flow of retransmission control by IP telephone control unit 112.

  If an ACK response cannot be received before a predetermined time has elapsed since the fifth packet retransmission at the MAC layer level, a retransmission-over event indicating a packet transmission error is generated in the MAC layer of the wireless communication unit 115. The This retry-over event is sent to the IP telephone control unit 112. The IP telephone control unit 112 determines that a packet transmission error has occurred, and proceeds to a retransmission control process. In this case, IP telephone control unit 112 suspends retransmission of the packet until the next packet transmission timing arrives. When the next packet transmission timing arrives, the IP telephone control unit 112 generates a single voice RTP packet by combining the data in the voice RTP packet immediately before the transmission error occurs and the voice data to be transmitted next. The voice RTP packet is transmitted.

  For example, when the voice RTP packet transmission interval is 20 ms, if a transmission error of the voice RTP packet (160 bytes) that the IP telephone control unit 112 tries to transmit at time T1 occurs, a retransmission over event is generated in the MAC layer of the wireless LAN. To the IP telephone control unit 112. Then, at the timing when the IP telephone control unit 112 transmits the voice packet at time T1 + 20 ms, the voice data that has just been over-retransmission over is combined and transmitted as a voice RTP packet (160 + 160 = 320 bytes).

  What is important here is the maximum number of combined packets. RTCP is also used in the IP telephone, and jitter information indicating jitter time is stored in the Interarrival Jitter of the reception report block of the RR packet from the other party, and the maximum combined packet number is determined using this. That is, assuming that the partner terminal has a jitter buffer of a size that can absorb at least jitter, the jitter buffer size of the partner terminal is estimated based on the jitter time (Interarrival Jitter) transmitted from the partner terminal.

  FIG. 6 shows the configuration of an RR packet defined in RFC3550. An item of Interarrival Jitter is defined in the reception report block of the RR packet, and includes the jitter time calculated on the receiving terminal side. The jitter time is the average deviation (standard absolute value) of the arrival interval difference between packets. For example, the first packet arrival interval (arrival time-transmission time) is 10 ms, the second packet arrival interval (arrival time-transmission time) is 20 ms, and the third packet arrival interval (arrival time-transmission time). Is 30 m, the average arrival interval is 20 ms, and the jitter time is 10 ms. A specific method for calculating the jitter time is described in p35 of the RFC3550 standard.

  Further, the round trip time (RTT) can be calculated using RTCP, and the jitter buffer of the partner terminal can be estimated based on the round trip time and the above jitter time.

  The round trip time (RTT) indicates the time required for a packet to reciprocate between the transmitting terminal and the receiving terminal. The round trip time (RTT) can be calculated using RTCP. A specific method for calculating the round trip time (RTT) is described in p40 of the RFC3550 standard.

  As described above, the maximum number of voice packets at the time of retransmission in the IP telephone control unit 112 must be within the jitter buffer size of the counterpart terminal. For example, if the counterpart terminal is equipped with an 800-byte jitter buffer, the IP telephone control unit 112 can transmit (retransmit) five voice packets together. However, even if RTCP is used, the jitter buffer size of the partner terminal cannot be known. Therefore, in this embodiment, the jitter buffer size is estimated from the above-described jitter time and round trip time. Hereinafter, a specific estimation method will be described.

  First, a case where the jitter buffer size is estimated based on both the round trip time and the jitter time will be described.

  When implementing the jitter buffer, the jitter buffer size may be dynamically changed by making it adaptive, but it may be possible to use the maximum jitter buffer size. Since the round trip timer is a required time for reciprocation, the time required for one direction is approximately halved, but it is considered that the jitter buffer on the receiving side is implemented for half of the round trip time. Therefore, the jitter buffer size is estimated as follows.

Jitter buffer size = (round trip time / 2 + jitter time) × 160 bytes / 20 ms
From the above, the maximum number of voice packets at the time of retransmission in the IP telephone control unit 112 is as follows by dividing the estimated jitter buffer size by the packet size (160 bytes).

Maximum number of voice packets at the time of retransmission = (round trip time / 2 + jitter time) / 20 + 1
Here, “+1” is used to round up the fraction.

  As described above, the IP telephone control unit 112 determines the round trip time by RTCP, and the total value of ½ of the round trip time and the jitter time is set to a value per unit time sent to the partner terminal by packet transmission. A value obtained by multiplying the data amount (160 bytes / 20 ms) can be estimated as the jitter buffer size.

In addition, a value obtained by multiplying the jitter time by the transmission data amount per unit time may be estimated as the jitter buffer size.
Jitter buffer size = jitter time × 160 bytes / 20 ms
It becomes.

  The above is a method for determining the maximum transmission packet size by estimating the jitter buffer size of the partner terminal from the RTCP RR packet from the partner terminal, but the private extensions SDES of the RTCP SDES (Source Description) A method is also possible in which the jitter buffer size is exchanged with the counterpart terminal using item to receive the jitter buffer size from the counterpart terminal so that the maximum number of packets to be combined is within the counterpart jitter buffer size. Private extensions SDES item of SDES (Source Description) is described in p45 of the RFC3550 standard.

  In addition to the method of estimating the jitter buffer size of the partner terminal from the jitter time and round trip time, or the method of receiving the jitter buffer size from the partner terminal using the Private extensions SDES item, the IP telephone control unit 112 There is also a method defined by the configuration information. That is, this is a method in which the jitter buffer size of the counterpart terminal is defined as 800 bytes in the configuration information file or the like. In the case of this method, since the partner terminal has various terminals, the expected minimum jitter buffer size of the connected terminal is defined. In other words, although the jitter buffer size varies depending on the type of terminal, the jitter buffer size can be roughly classified into several ways. By registering the lowest jitter buffer size in the terminal 11 beforehand, The maximum transmission packet size can be limited based on the jitter buffer size.

  Next, the procedure of voice packet transmission processing executed by the terminal (A) 11 will be described with reference to the flowchart of FIG.

  The IP telephone control unit 112 of the terminal (A) 11 transmits a packet addressed to the partner terminal to the wireless network by transmitting a voice RTP packet to the wireless communication unit 115 at a time interval of once every 20 ms (step S11). . When this packet is transmitted normally, that is, when an ACK response is returned from the wireless router 12 within the maximum number of retransmissions at the MAC layer level and a retransmission number over event does not occur (NO in step S12), the terminal ( A) The IP telephone control unit 112 of 11 sends the next voice RTP packet to the wireless communication unit 115 at the next packet transmission timing, thereby sending the next packet addressed to the partner terminal to the wireless network (step S13). , S14).

  If the packet is not normally transmitted, that is, if the ACK response is not returned from the wireless router 12 within the maximum number of retransmissions at the MAC layer level and a retransmission number over event occurs (YES in step S12), the terminal ( A) The IP telephone control unit 112 of 11 holds the retransmission of the packet in which the transmission error has occurred until the next packet transmission timing (step S15). When the next packet transmission timing arrives (YES in step S16), the IP telephone control unit 112 combines the voice data of the voice RTP packet immediately before the packet transmission has failed and the next voice data to be transmitted. One voice RTP packet is generated (step S17). The IP telephone control unit 112 determines whether or not the data size of the generated voice RTP packet is within the jitter buffer size of the counterpart terminal (step S18).

  If the data size of the generated voice RTP packet is within the jitter buffer size of the counterpart terminal (YES in step S18), the IP telephone control unit 112 sends the generated voice RTP packet to the wireless communication unit 115, The next packet addressed to the partner terminal is sent to the wireless network (step S19).

  On the other hand, if the data size of the generated voice RTP packet exceeds the jitter buffer size of the counterpart terminal (NO in step S18), the IP telephone control unit 112, for example, has the earliest 160-byte voice data in time. Packet size adjustment processing such as deleting a part from the voice RTP packet is executed to limit the data size of the voice RTP packet to be equal to or smaller than the jitter buffer size of the counterpart terminal (step S20). Then, the IP telephone control unit 112 sends the next packet addressed to the counterpart terminal to the wireless network by sending it to the wireless communication unit 115 that has been subjected to the packet size adjustment processing and is a voice RTP packet (step S21).

  Note that the packet size adjustment processing is not limited to deleting only the 160-byte audio data portion that is older in time, and it is only necessary to delete any 160-byte audio data portion.

  Next, a process for limiting the maximum packet size based on the jitter buffer size of the counterpart terminal will be described with reference to the flowchart of FIG.

  The IP telephone control unit 112 detects the jitter buffer size of the partner terminal through communication with the partner terminal (step S21). Specifically, the IP telephone control unit 112 estimates the jitter buffer size of the partner terminal based on information such as jitter time or round trip time obtained by communication with the partner terminal, or SDES (Source Description) The jitter buffer size of the partner terminal is determined by the process of exchanging the jitter buffer size with the partner terminal using Private extensions SDES item.

  Next, IP telephone control section 112 defines the determined jitter buffer size of the partner terminal as the maximum data size (maximum number of packets combined) of voice RTP packets that can be transmitted (step S22). Then, in steps S18 and S20 of FIG. 7 described above, a process of limiting the retransmission packet size to a specified maximum data size or less is executed (step S23).

  As described above, in the present embodiment, when retransmission over in the wireless communication unit 115 occurs, retransmission is not performed immediately, but retransmission is suspended until the next audio data transmission timing, and then the next audio data transmission timing is reached. When the voice data arrives, the wireless communication unit 115 transmits the voice data in which the over-retransmission occurs in combination with the next voice data, thereby transmitting the voice data in real time without imposing unnecessary load on the CPU or the network. It becomes possible. Further, by limiting the maximum packet size at the time of retransmission to be equal to or smaller than the jitter buffer size of the counterpart terminal, it is possible to recover even when a burst error (continuous retransmission over error) occurs.

  In this embodiment, the case where audio data is transmitted to the counterpart terminal in real time has been described as an example. However, the retransmission control processing of the present embodiment is also applicable to the case where video data is transmitted to the counterpart terminal in real time. Can do. The system of this embodiment may be realized as a wireless LAN IP videophone system for transmitting both audio and video to a partner terminal in real time via a wireless network. In this case, audio data transmission In addition, the retransmission control process of this embodiment may be executed individually for each transmission of video data.

  Further, the present invention is not limited to the above-described embodiment as it is, and can be embodied by modifying the constituent elements without departing from the scope of the invention in the implementation stage. In addition, various inventions can be formed by appropriately combining a plurality of components disclosed in the embodiment. For example, some components may be deleted from all the components shown in the embodiment. Furthermore, constituent elements over different embodiments may be appropriately combined.

The block diagram which shows the structural example of the data transmission system using the data transmission terminal device which concerns on one Embodiment of this invention. The block diagram which shows the function structure of the data transmission terminal device of the embodiment. The figure which shows the example of the packet transmission operation | movement performed by the data transmission terminal device of the embodiment. The figure for demonstrating the resending operation | movement performed by the radio | wireless communication part provided in the data transmission terminal device of the embodiment. The figure for demonstrating the resending operation | movement performed by the IP telephone control part provided in the data transmission terminal device of the embodiment. The figure for demonstrating the RR packet which the data transmission terminal device of the embodiment receives from a partner terminal. 6 is an exemplary flowchart for explaining the procedure of packet retransmission control processing executed by the data transmission terminal device of the embodiment; 9 is a flowchart for explaining processing for limiting the maximum packet size based on the jitter buffer size of the counterpart terminal, which is executed by the data transmission terminal device of the embodiment.

Explanation of symbols

  DESCRIPTION OF SYMBOLS 11 ... Data transmission terminal device, 12, 14 ... Wireless router, 13 ... Network, 15 ... Remote terminal, 111 ... Voice input part, 112 ... IP telephone control part, 113 ... RTP transmission process part, 114 ... IP transmission process part, 115: A wireless communication unit.

Claims (10)

A data transmission terminal device for transmitting audio or video in real time to a partner terminal via a wireless network,
Packet transmitting means for transmitting a packet of a predetermined size including the audio or video data to the counterpart terminal via the wireless network at a predetermined time interval;
When a transmission error occurs for a packet sent to the wireless network, the retransmission of the packet in which the transmission error has occurred is suspended until the next packet transmission timing arrives, and the transmission error occurs at the next packet transmission timing Retransmission control means for generating one packet by combining the data in the immediately preceding packet and the next data to be transmitted, and transmitting the generated packet to the counterpart terminal via the wireless network A data transmission terminal device.   The maximum packet size generated by the retransmission control means when a packet transmission error occurs continuously at a plurality of consecutive packet transmission timings is limited to a buffer size for jitter absorption of the counterpart terminal or less. The data transmission terminal apparatus according to claim 1, wherein:   Further comprising buffer size determining means for determining a buffer size for jitter absorption of the counterpart terminal by communication with the counterpart terminal, the maximum packet size generated by the retransmission control means is equal to or less than the determined buffer size The data transmission terminal device according to claim 2, wherein the data transmission terminal device is limited to:   The buffer size determining means estimates, as the buffer size, a value obtained by multiplying a jitter time transmitted from the partner terminal by a data amount per unit time transmitted to the partner terminal by transmission of the packet. 4. The data transmission terminal device according to claim 3, further comprising means.   The buffer size determining means determines a round trip time required for a packet to reciprocate between the data transmission terminal device and the counterpart terminal, and ½ of the round trip time is transmitted from the counterpart terminal. The means for estimating the buffer size includes a value obtained by multiplying a total value with a jitter time by a data amount per unit time transmitted to the counterpart terminal by transmission of the packet. 3. The data transmission terminal device according to 3. If an acknowledgment for a packet sent to the wireless network is not returned from a router in the wireless network, further comprising means for repeating retransmission of the packet until the acknowledgment is returned;
The data transmission terminal apparatus according to claim 1, wherein the retransmission control means determines that a transmission error has occurred for the packet when the number of retransmissions of the packet exceeds a predetermined value. A data transmission terminal device for transmitting audio or video in real time to a partner terminal via a wireless network,
Packet transmitting means for transmitting a packet of a predetermined size including the audio or video data to the counterpart terminal via the wireless network at a predetermined time interval;
If an acknowledgment for a packet sent to the wireless network is not returned from a router in the wireless network, the packet is retransmitted until the acknowledgment is returned, and the number of retransmissions of the packet exceeds a predetermined value. A means for generating a retransmission count over event indicating that a transmission error has occurred for the packet;
When the retransmission number over event occurs, the retransmission of the packet in which the transmission error has occurred due to the retransmission number excess being suspended until the next packet transmission timing arrives, and at the next packet transmission timing, the transmission error due to the retransmission number exceeding Retransmission control means for generating one packet by combining the data in the packet immediately before the occurrence of the error and the next data to be transmitted, and transmitting the generated packet to the counterpart terminal via the wireless network; ,
Means for determining a buffer size for jitter absorption of the counterpart terminal by communication with the counterpart terminal;
The data transmission terminal apparatus, wherein the maximum packet size generated by the retransmission control means is limited to be less than or equal to the determined buffer size. A data transmission method for transmitting audio or video in real time to a partner terminal via a wireless network,
A packet transmission step of transmitting a packet of a predetermined size including the audio or video data to the counterpart terminal via the wireless network at a predetermined time interval;
When a transmission error occurs for a packet sent to the wireless network, the retransmission of the packet in which the transmission error has occurred is suspended until the next packet transmission timing arrives, and the transmission error occurs at the next packet transmission timing A retransmission control step of generating one packet by combining the data in the immediately preceding packet and the next data to be transmitted, and transmitting the generated packet to the counterpart terminal via the wireless network. A data transmission method.   A limiting step of limiting a maximum packet size generated by the retransmission control step when a packet transmission error occurs continuously at a plurality of continuous packet transmission timings to a jitter absorption buffer size or less possessed by the counterpart terminal The data transmission method according to claim 8, further comprising:   A buffer size determining step of determining a buffer size for jitter absorption of the counterpart terminal by communication with the counterpart terminal; and the limiting step sets the maximum packet size to be equal to or less than the determined buffer size. The data transmission method according to claim 9, wherein the data transmission method is limited.

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