KR20160026668A - APPARATUS FOR CONTROLLING SIZE OF VoIP PACKET AND METHOD THEREOF - Google Patents

Abstract

The present invention relates to a device and a method for controlling a VoIP packet size and, more particularly, to a device and a method for controlling a VoIP packet size, in which a receiving terminal for receiving a VoIP packet in real time measures end-to-end transmission delay information, jitter information, and packet loss rate and transmits the measured end-to-end transmission delay information, jitter information, and packet loss rate to a transmission terminal, and the transmission terminal generates an App-VoIP PDU by adjusting the size of the App-VoIP PDU to a maximum size within a range that does not exceed maximum allowable transmission delay time, jitter value, and packet loss rate for ensuring differentiated service quality based on the received end-to-end transmission delay information, jitter information, and packet loss rate.

Description

[0001] APPARATUS FOR CONTROLLING SIZE OF VOIP PACKET AND METHOD THEREOF [0002]

The present invention relates to an apparatus and method for adjusting a VoIP packet size of a Voice over Internet Protocol (VoIP) communication network, and more particularly to a method and apparatus for adjusting a VoIP packet size of a VoIP packet, A VoIP packet size control apparatus and method for providing a terminal delay propagation delay information, a jitter information, and a packet loss rate on the basis of an end-to-end propagation delay information, an adaptation of a size of an App-VoIP PDU (application layer VoIP protocol data unit) .

Recently, VoIP (Voice over IP) -based real-time Internet voice communication services such as Skype (VoIP) and Kakaotalk Voicetalk have been widely used in portable terminals such as smart phones, smart pads, .

The transmission of the VoIP voice information is performed by generating a VoIP voice information in a PCM (Pulse Coded Modulation) method, and then transmitting the VoIP voice information through an application layer protocol data unit (App-PDU) ) Is generated and transmitted in a separate packet, it is transmitted through the Real-time Transport Protocol (RTP) / User Datagram Protocol (UDP) / Internet Protocol (IP) .

That is, the application-layer protocol data unit (App-PDU) in the protocol layer of the VoIP service has a predetermined interval (for example, the sampling interval is 125 占 퐏 for sampling at 8000 times per second) A small sampled isochronous App-PDU is generated. The generated App-PDU is transmitted through the protocol structure of App-VoIP, RTP, UDP and IP sequentially.

The size of the App-VoIP PDU at the transmitter corresponds to the amount of VoIP voice information generated at intervals of 10 ms to 100 ms. When VoIP voice information is generated by the 64 Kbps PCM method, 8 bytes per 1 ms are generated, and voice data of 80 bytes per 10 ms, 800 bytes per 100 ms, and 8000 bytes per second is generated.

When the voice data compression function such as G.729, enhanced variable rate codec (EVRC) or adaptive multi-rate (AMR) is used, the size of the App-VoIP PDU is reduced to 10 to 30 Bytes per 10 to 20 ms.

In the App-VoIP / RTP / UDP / IP protocol structure used for delivering VoIP voice information, the size of the protocol header for each layer is at least 44 bytes (16 bytes for the RTP header, 8 bytes for the UDP header, 20 bytes ). Considering the overhead of this protocol structure, it can be said that keeping the size of App-VoIP PDU as large as possible is advantageous also in terms of packet transmission performance.

However, in the case of real-time multimedia service, in order to guarantee the required quality of service, the end-to-end propagation delay should be maintained within 150ms, and in order to satisfy the condition that the jitter should be maintained within 50ms, Should be limited.

The end-to-end delay of the VoIP voice information is a time taken from when the App-VoIP packet is generated in the transmitting terminal to when it is transmitted to the receiving terminal. The end-to-end delivery delay of VoIP voice information includes a delay due to traffic congestion in a subscriber access network and an intermediary transmission network. Therefore, the VoIP voice information is not maintained at a constant value, but has a characteristic of dynamically varying according to traffic conditions and time of a communication network.

At present, in the case of the implemented VoIP transmitting terminal, the size of the App-PDU can be specified in the program initial setting. However, the size of the App-PDU specified in the initial setting is not changed appropriately according to the network traffic condition and the available bandwidth, but is maintained at a fixed value. That is, since the conventional VoIP transmitting terminal generates the App-PDU size in a fixed size irrespective of the position of the VoIP transmitting / receiving terminal and the delay time of the relay transmission network, the available bandwidth is wasted and packet processing performance is inefficient, Problems arise.

Korean Registered Patent No. 1295707 (Registered on Mar. 08, 06)

SUMMARY OF THE INVENTION It is an object of the present invention to provide a VoIP packet receiving terminal, which measures the end-to-end transmission delay information, the jitter information and the packet loss rate, And the size of the App-VoIP PDU within the range in which the transmitting terminal does not exceed the maximum allowable propagation delay time, the jitter value and the packet loss rate for guaranteeing the service quality based on the end-to-end propagation delay information, the jitter information, And adjusting the size of the VoIP packet to a maximum size.

It is another object of the present invention to provide a real-time service that allows less than a certain range of packets to be delivered during each periodic interval to exceed the packet delivery capability (end-to-end delivery delay information, jitter information and packet loss rate value) Based on the end-to-end delivery delay information, the jitter information, and the packet loss rate measured and provided from the receiving terminal, the differentiated service quality level is further classified into the service quality class differentiated according to the over- A VoIP packet size control device and method for generating a packet by adjusting the size of an App-VoIP PDU to a maximum size within a range satisfying a packet transmission performance (end-to-end transmission delay information, jitter information and packet loss rate) .

According to an aspect of the present invention, there is provided an apparatus for controlling a VoIP packet size according to an embodiment of the present invention includes a transmitting terminal using a VoIP real-time voice information service and a receiving terminal, And a packet loss rate, and transmits the measurement result to the transmitting terminal. The transmitting terminal calculates a maximum allowable transmission delay time, a jitter value, and a packet loss rate for ensuring quality of service based on the end-to-end transmission delay information, jitter information, The size of the VoIP application layer protocol data unit (App-VoIP PDU) is adjusted to the maximum size to generate the VoIP packet.

According to another aspect of the present invention, there is provided an apparatus for controlling a VoIP packet size, the apparatus including a transmitting terminal using a VoIP real-time voice information service and a receiving terminal, An RTP / RTCP layer for measuring and transmitting information, jitter information and packet loss rate; And a maximum allowable propagation delay time for guaranteeing service quality based on the end-to-end propagation delay information, jitter information, and packet loss rate received by the RTP / RTCP layer, a jitter value and a packet loss rate, And generates a protocol data unit (App-VoIP PDU) by adjusting its size to the maximum size.

According to another aspect of the present invention, there is provided a method for adjusting a VoIP packet size, the method comprising: receiving end-to-end delivery delay information, jitter information, and packet loss rate from a receiving terminal that receives a VoIP packet in real- And a maximum permissible transmission delay time, a jitter value and a packet loss rate for guaranteeing a differentiated service quality based on the end-to-end transmission delay information, the jitter information and the packet loss rate, And a second step of adjusting the size of the VoIP application layer protocol data unit (App-VoIP PDU) to a maximum size to generate a VoIP packet.

The present invention generates the App-VoIP PDU by adjusting the size of the App-VoIP PDU to the maximum size based on the end-to-end transmission delay information, the jitter information, and the packet loss rate measured and transmitted from the receiving terminal, It is possible to transmit the packet by reflecting the change of the communication network state such as the delay time of the transmission network and the available bandwidth in real time, thereby improving the VoIP packet processing performance and transmission efficiency.

1 is a block diagram of a VoIP packet size control apparatus according to the present invention;
FIG. 2 is a flowchart illustrating a method for measuring a transmission delay information, jitter information, and packet loss rate, and providing the same to a transmitting terminal according to the present invention.
3 is a flowchart illustrating a method of adjusting the size of an App-VoIP PDU of a transmitting terminal according to the present invention.

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

Considering the fact that the end-to-end delivery delay is maintained within 150ms and the jitter is maintained within 50ms, there is no problem in providing quality of service, in the case of real-time multimedia service in the present invention, It is suggested that the maximum size can be generated within a limit that satisfies the condition that the delay of 150 ms between the transmission and the jitter is maintained within 50 ms.

In the present invention, when the VoIP voice information is transmitted and received between the transmitting and receiving terminals in real time, the receiving terminal measures the total delay time including the de-jittering buffer delay and the packet loss rate, and outputs the result as RTP / To the transmitting terminal. The transmitting terminal may vary the size of the App-PDU to the maximum size within a range satisfying the quality of service based on the received result.

1 is a block diagram of a VoIP packet size control apparatus according to the present invention. As shown in FIG. 1, the VoIP packet size control apparatus according to the present invention includes a transmitting terminal 10 and a receiving terminal 20 using a VoIP real-time voice information service.

The receiving terminal 20 measures the end-to-end transmission delay information, the jitter information, and the packet loss rate, and transmits the measurement result to the transmitting terminal 10. Based on the transmitted end-to-end delivery delay information, the jitter information, and the packet loss rate, the transmitting terminal 10 sets a maximum allowable transmission delay time, a maximum allowed jitter value, and a maximum allowable packet loss rate A VoIP packet is generated by adjusting the size of a VoIP application layer protocol data unit (App-VoIP PDU) to a maximum size.

The transmitting terminal 10 and the receiving terminal 20 are connected to the App-VoIP layer 11 21, the RTP / RTCP layer 12 22, the UDP layer 13 23 and the IP layer 14 24).

The transmitting terminal 10 and the receiving terminal 20 are connected to the transmitting subscriber access network 30 and the receiving subscriber accessing network 30. The transmission side subscriber access network 30 and the reception side subscriber access network 30 are connected through one or more relay transmission networks 40. The one or more relay transmission networks 40 include a relay WAN (Transit WAN).

The subscriber access network 30 may include, for example, Ethernet, a wide area network (WAN), a wireless LAN, a wireless WAN, and the like, but is not limited thereto.

Voice information (App-VoIP PDU) generated in the App-VoIP layer (VoIP application layer) 11 of the transmitting terminal 10 includes App-VoIP 11, RTP 12, UDP 13 and IP 14 ), And then transmitted to the receiving terminal 20 by the lower data communication networks 30 and 40. [

The RTP / RTCP protocol provides an end-to-end real-time information transfer function in the App-VoIP service. The RTP / RTCP layer 22 measures the end-to-end transfer delay information, the jitter information and the packet loss rate To the transmitting terminal (10).

The RTP / RTCP layer 12 of the transmitting terminal 10 receives the end-to-end propagation delay information, the jitter information and the packet loss rate transmitted from the receiving terminal 20 and transmits the received information to the App-VoIP layer 11. The App-VoIP layer 11 determines whether the App-VoIP layer 11 is within the range of the maximum allowable propagation delay time, jitter value, and packet loss rate for ensuring quality of service, based on the end-to-end propagation delay information, jitter information, And adjusts the size of the VoIP PDU to the maximum size.

In case of a real-time multimedia service, the maximum allowable transmission delay time should be maintained within 150 ms, and the jitter value should be maintained within 50 ms.

The end-to-end delivery delay of the VoIP voice information is a time taken from when the packet is generated in the transmitting terminal 10 to when the packet is transmitted to the receiving terminal 20. The end-to-end delivery delay of VoIP voice information includes a waiting delay according to the App-PDU size, a transmission side subscriber access network 30 delay, a relay transmission network 40 delay, a reception side subscriber access network 30 delay, A de-jittering buffer delay at the receiving terminal 20, and the like.

Jitter represents a change caused by an end-to-end delay of transmission and is used for communication between a transmitting terminal 10 and a receiving terminal 20 in a storage medium, a transmission channel, an analog-to-digital converter (ADC) Frame collision and retransmission in the transmission equipment and the wireless LAN access network, and packet forwarding delay in the IP packet router, which are important factors for the deterioration of digital signal quality and system performance. Accordingly, the receiving terminal 20 reduces the influence of jitter caused when the voice packets are transmitted from the transmitting terminal 10 to the receiving terminal 20 through the wired / wireless communication networks 30 and 40, A de-jittering buffer is used to process the packet.

FIG. 2 is a flowchart illustrating a method in which a receiving terminal 20 according to the present invention measures end-to-end transmission delay information, jitter information, and packet loss rate and provides the measured transmission delay information to the transmitting terminal 10.

The receiving terminal 20 calculates an end-to-end delay value based on the RTP time stamp of a new packet each time a new App-VoIP / RTP packet arrives. That is, the receiving terminal 20 compares the transmission time (generation time) recorded in the RTP time stamp of the new packet with the arrival time of the new packet to calculate the end-to-end transmission delay information (S10) (S12).

The receiving terminal 20 compares the RTP time stamp of the arrived new packet with the RTP time stamp of the previously arrived packet to calculate jitter information (S14).

In addition, the receiving terminal 20 checks whether the packet is lost by using the RTP sequence number of the arrived new packet (S16).

The receiving terminal 20 performs the steps S10-S16 at predetermined time intervals (for example, one second) or every predetermined number of times (S18).

Thus, the receiving terminal 20 periodically measures the end-to-end transmission delay information, the jitter information, and the packet loss rate each time a predetermined number of packets are received or set periodically, and provides the measurement result to the transmitting terminal 10 (S20) S22). The jitter information may include a jitter maximum value, a minimum jitter value, and a jitter average value. The receiving terminal 20 may calculate the jitter maximum value, the minimum jitter value, and the jitter average value in the case of a predetermined number of packet reception or a set period, and may transmit the jitter maximum value, the jitter minimum value, and the jitter average value to the transmitting terminal 10.

In another embodiment, the receiving terminal 20 may determine that the measured end-to-end delivery delay, jitter information, and packet loss rate are less than the measured end-to-end delivery delay only if there is a change in at least one of the measured end- Information, jitter information, and packet loss rate to the transmitting terminal 10. [

3 is a flowchart illustrating a method of adjusting the App-VoIP PDU size of the transmitting terminal 10 according to the present invention.

The transmitting terminal 10 periodically generates a voice signal sample and prepares it for transmission in an RTP packet. At this time, the transmitting terminal 10 records the generation time (transmission time) of the voice signal samples in the time stamp included in the RTP packet header and transmits the recording time (S30).

Thereafter, until the arrival of the end-to-end delivery delay information, the jitter information (maximum value, minimum value, average value) and the packet loss rate of the VoIP / RTP packet measured from the receiving terminal 20 to the transmitting terminal 10, The VoIP / RTP packet is continuously transmitted (S32).

When the end-to-end delivery delay information, the jitter information (maximum value, minimum value, average value) and the packet loss rate of the VoIP / RTP packet measured from the receiving terminal 20 are received, the transmitting terminal 10 transmits, VoIP packets are generated by adjusting the size of the App-VoIP PDU including the voice signal samples to the maximum size within a range not exceeding the maximum allowable propagation delay time for guaranteeing quality and the jitter value and the packet loss rate (S34).

Also, the transmitting terminal 10 is operated to dynamically select and use one of various error recovery techniques considering the received end-to-end transmission delay information, jitter information (maximum value, minimum value, average value), and packet loss rate (S36) .

When a bit error occurs in the App-VoIP PDU, the receiving terminal 20 can solve the bit error problem according to a forward error correction (FEC) technique, and the received end-to-end propagation delay information, jitter information, If the packet loss rate is satisfied with the service quality level, the transmitting terminal 10 can select and use a forward error correction (FEC) technique among various error recovery techniques. In such a case, maintaining the maximum App-VoIP PDU size as much as possible helps improve the performance of the network.

However, when the bit error processing is performed using the selective MPDU re-transmission technique without using the FEC technique, if the size of the App-VoIP PDU is large, the probability of occurrence of a frame error becomes high, It is advantageous that a frame of size is maintained. Therefore, adjusting the size of the App-VoIP PDU considering the bit error rate of the transmission channel further improves the packet transmission performance.

Meanwhile, in the VoIP packet size control apparatus according to the present invention, when a predetermined range (e.g., 5% or 10%) of the packets transmitted in units of a predetermined time exceeds the specified transmission delay information, jitter information, It is possible to further classify and manage the allowed real-time services into different service quality classes according to the over-allowed range intervals.

In this case, in the case of the real-time service corresponding to each service quality class classified according to the over-permissible range intervals, the transmitting terminal 10 calculates the transmission delay time, the jitter information, and the packet loss rate based on the received end- The App-VoIP PDU size may be adjusted to the maximum size so as to satisfy the over-allowed range.

In addition, the transmitting terminal 10 may determine whether one of the measured jitter maximum value, jitter minimum value, and jitter average value transmitted from the receiving terminal 20 is selected according to the real-time multimedia service quality level and satisfies the satisfaction. If the quality of service level is to be met very tightly, the sending terminal 10 may adjust App-VoIP PDU size based on whether a jitter maximum is met.

On the other hand, when the VoIP transmitting / receiving terminals 10 and 20 are located on the same campus, the relay transmission network delay is maintained within approximately 2 ms, so that the VoIP App-PDU can be maintained as large as about 100 ms. As another example, when the VoIP transmitting / receiving terminals 10 and 20 are located in the international (e.g., Korea-US), since the relay transmission network delay is about 40 to 80 ms and the VoIP App-PDU is relatively large, ms. < / RTI > Also, since the end-to-end transmission delay includes a delay due to traffic congestion in the subscriber access network and the relay transmission network, the end-to-end transmission delay is not maintained at a constant value but has a characteristic of dynamically varying with time.

Therefore, the receiving terminal 20, which receives VoIP packets in real time, measures the end-to-end transmission delay information, the jitter information and the packet loss rate and transmits them to the transmitting terminal 10, The App-VoIP PDU size is adjusted to a maximum size within a range in which the maximum allowable propagation delay time, jitter value, and packet loss rate are not exceeded to guarantee service quality based on delay information, jitter information, and packet loss rate, The size of the voice packet can be adjusted to the maximum size by reflecting the location of the transmitting terminal 10 and the receiving terminal 20 and the state of the communication network in real time.

10: transmitting terminal 20: receiving terminal

Claims (11)

A transmitting terminal using a VoIP real-time voice information service and a receiving terminal,
The receiving terminal measures end-to-end transmission delay information, jitter information, and packet loss rate, and transmits the measurement result to the transmitting terminal,
The transmitting terminal transmits the VoIP application layer protocol data, the jitter information, and the packet loss rate within a range that does not exceed the maximum allowable propagation delay time, jitter value, and packet loss rate for ensuring service quality based on the received end- Unit (App-VoIP PDU) is adjusted to a maximum size to generate a VoIP packet.
The method according to claim 1,
Wherein the receiving terminal periodically measures the end-to-end transmission delay information, the jitter information, and the packet loss rate every time a predetermined number of packets are received or transmitted, and transmits the measurements to the transmitting terminal.
The method according to claim 1,
The receiving terminal measures the measured end-to-end delivery delay information, the jitter information, and the packet loss rate only when at least one of the measured end-to-end delivery delay information, the jitter information, and the packet loss rate is changed more than a predetermined value, To the transmitting terminal.
The method according to claim 1,
Wherein the transmitting terminal is operated to dynamically select and use one of various error recovery techniques according to the value of the received end-to-end transmission delay information, the jitter information, and the packet loss rate.
The method according to claim 1,
For real-time services that are allowed to exceed specified delivery delay information, jitter information, and packet loss rate values,
Differentiated service quality class according to the over allowable range,
In the case of a real-time service corresponding to each of the classified service quality levels, the transmitting terminal determines the size of the App-VoIP PDU so as to satisfy the respective over-allowed range intervals based on the received end-to-end transmission delay information, jitter information, To a maximum size.
6. The method of claim 5,
The jitter information includes a jitter maximum value, a jitter minimum value, and a jitter average value, which are calculated for each set time period or each set time period,
Wherein the transmitting terminal selects one of the jitter maximum value, the minimum value, and the average value according to a real-time multimedia service quality level including the classified service quality level to determine whether the jitter is satisfied or not.
The method according to claim 1,
Wherein the transmitting terminal receives the end-to-end transmission delay information, the jitter information, and the packet loss rate from the receiving terminal through the RTP / RTCP protocol.
A transmitting terminal using a VoIP real-time voice information service and a receiving terminal,
The transmitting terminal and the receiving terminal,
An RTP / RTCP layer for measuring and transmitting end-to-end transmission delay information, jitter information, and packet loss rate; And
A maximum allowable propagation delay time, a jitter value, and a packet loss rate for ensuring quality of service based on the end-to-end propagation delay information, the jitter information, and the packet loss rate received by the RTP / And an App-VoIP layer for adjusting the size of a data unit (App-VoIP PDU) to a maximum size.
A first step of receiving end-to-end delivery delay information, jitter information, and packet loss rate transmitted from a receiving terminal that receives a VoIP packet in real time via a VoIP communication network; And
The transmitting terminal transmits the VoIP application layer protocol data unit (PDU) within a range that does not exceed the maximum allowable propagation delay time, jitter value and packet loss rate for guaranteeing the service quality based on the received end-to-end propagation delay information, jitter information, (App-VoIP PDU) to a maximum size to generate a VoIP packet.
10. The method of claim 9,
The second step comprises:
VoIP PDU's to meet the over-allowed ranges allowed to exceed the set propagation delay information, jitter information and packet loss rate values based on the end-to-end propagation delay information, jitter information and packet loss rate received by the transmitting terminal And adjusting the size to a maximum size to generate a packet.
11. The method of claim 10,
The received jitter information includes a jitter maximum value, a minimum jitter value, and a jitter average value calculated every time a predetermined number of packets are received or every set period,
Wherein the transmitting terminal selects and applies one of the jitter maximum value, the minimum value and the average value according to a differentiated quality of service class satisfying the over-allowed range intervals.

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