KR100486447B1 - METHOD FOR CONTROLLING QoS FOR USER'S MOBILITY SUPPORT IN THE INTERNET MULTIMEDIA COMMUNICATION - Google Patents

Abstract

The rate control scheme for multimedia services in the Internet is designed to solve only the degradation of service quality caused by network overload in fixed networks. These techniques mistake the large packet loss that occurs temporarily when a user moves from one network to another in the mobile network as being caused by overcrowding, thereby reducing the transmission rate even though the user can be served at the original rate. This has the disadvantage of not being able to provide adequate quality of service and of effectively utilizing the available network resources. The present invention records the packet loss caused by the user's movement and does not include this value in the quality of service feedback information. This allows the server to be unaware of packet loss due to user movement. As a result, the server can control the transmission rate only for the degradation of the service quality due to network overcrowding, thereby providing the user with more appropriate quality of service. In addition, it prevents the problem of wasting available network resources by reducing the transmission rate due to network overcrowding that does not occur.

Description

METHOD FOR CONTROLLING QoS FOR USER'S MOBILITY SUPPORT IN THE INTERNET MULTIMEDIA COMMUNICATION}

The present invention relates to a method of controlling a quality of service (QoS) for guaranteeing user mobility in the internet multimedia communication. In particular, the present invention relates to a decrease in service quality due to network overcrowding when the quality of service is degraded. The present invention relates to a method of controlling the quality of service by classifying whether the quality of service is deteriorated.

TCP-Friendly Rate Control (hereinafter, abbreviated as TFRC) has been proposed to provide multimedia services with appropriate quality for the network condition in the Internet which does not guarantee end-to-end quality of service. TFRC is similar to the network congestion control technique of the Transmission Control Protocol (hereinafter referred to as TCP). When the service quality decreases, the TFRC assumes that the congestion of the service is caused by network congestion. (drop down / slow recovery) controls the rate. This method allows time for the overcrowded network to recover to its normal state.

When the multimedia service is applied to a mobile network, there is a problem of deterioration of service quality caused by instability of the wireless network connection and mobility of the user, in addition to network overcrowding. These degradations are caused by packet loss, which is characterized by a sudden, large amount of transients. Sudden transients means that it takes no time to recover to the original data reception state after a packet loss. Therefore, if the existing solution for solving network overcrowding is applied to the above quality factors, the mobile host has to bear the low transmission rate that the server transmits during the network overcrowding recovery time. In order to prevent such unnecessary degradation of service quality, the rate control scheme for overcrowding the network should not detect the degradation of service quality caused by the instability of the wireless network and the mobility of users.

To this end, in the TCP field, a method for distinguishing a service quality deterioration due to wireless network and mobility from a service deterioration due to network overcrowding has been proposed. First of all, packet loss caused by unstable wireless network has been proposed to solve the problem by placing a router (Mobility Support Router) (hereinafter referred to as MSR) that guarantees the mobility of the user at the interface between the fixed network and the mobile network. Since the MSR establishes a session with the fixed host on behalf of the mobile host, the fixed host actually sees only the image of the mobile host in the MSR, not the mobile host in the mobile network. As a result, the service quality degradation problem caused by the instability of the mobile network is managed only by the MSR, and the fixed host is only concerned with the service quality degradation problem due to network overcrowding in the fixed network. In addition, when the mobile host moves to another network, the MSR of the new network passes all the information about the session to support the role of hiding the user's mobility to the fixed host. Similarly, in the "non-TCP field", a method of reporting a quality of service in a multimedia session by providing a "RTP Monitoring Agent" (hereinafter referred to as RMA) at the same location is proposed. As a result, the multimedia server receives a report on the quality of service from the client and the RMA, respectively, to distinguish whether the packet loss is caused by network congestion in the fixed network or incompleteness of the mobile network.

In the case of user mobility, TCP proposes a technique for retransmitting lost packets without delay by the sender by informing the sender of the fact that the user moves from one network to another.

The rate control scheme proposed for the conventional multimedia service does not reflect the mobility of the user, that is, the packet loss caused by the handoff, by dealing only with network overcrowding. Packet loss caused by handoff is characterized by temporary mass occurrence, unlike packet loss caused by network overcrowding. This means that a mobile host can be served at the existing transmission rate immediately after packet loss occurs. However, the solution to the existing network overcrowding is that the server reduces the transmission rate after the handoff, so that the mobile host is provided with a lower quality of service than it can receive and does not use the resources of the available network efficiently. Has its drawbacks.

The present invention has been made to solve the above-described drawbacks, and does not report to the server about the packet loss caused by the user's network movement while receiving the multimedia service. The purpose of the present invention is to provide a service quality control method for guaranteeing user mobility in Internet multimedia communication by improving the deterioration of service quality caused by unnecessarily lowering a transmission rate and by efficiently using available network resources.

The present invention for achieving the above object, in the Internet multimedia communication system with a handoff function, the user hides the information on the packet loss caused by his handoff and the information on the quality of service that he is receiving to the server A method for reporting, comprising: a first step of the user corresponding to a mobile host detecting his or her handoff in a mobile IP network provided in the internet multimedia communication system; A second step of the mobile host collecting information related to a quality of service received by the mobile host while performing the sensed handoff procedure; A third step of calculating the number of packets lost during the handoff using the collected information during the handoff period; And a fourth step of correcting "cumulative number of packets lost" and "fraction lost" values, which are RTCP quality of service fields related to packet loss, by using the number of lost packets during the handoff period.

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

In the present invention, it is possible to flexibly provide appropriate service quality for user mobility in addition to network congestion by targeting a rate control scheme using a real-time transport control protocol (hereinafter, referred to as RTCP). Way. RTCP was designated the Request for Comments (RFC) standard by the Internet Engineering Task Force (IETF) in 1996 (RFC 1889).

RTCP is primarily responsible for the number of packets lost since the beginning of the session for the quality of service that each host participating in the multimedia session, and the rate of packet loss since the previous RTCP packet was sent. It is responsible for reporting to the server using lost and interarrival jitter values. Currently, the rate control scheme for multimedia services allows the server to change the rate using this RTCP feedback information.

If a user disconnects the network temporarily because of a handoff in the mobile network, packet loss occurs, and the result is recorded in the user's RTCP feedback information and reported to the streaming server. Streaming servers use this information to misunderstand that overcrowding has occurred. In addition, by unnecessarily overcrowding the network according to the TFRC algorithm, users experience low quality of service due to reduced data rates. In addition, users will receive this low quality of service until the streaming server returns to its original rate. In order to provide an appropriate quality of service in consideration of the mobility of the user, the present invention introduces the concept of fast recovery. This hides the packet loss caused by the user's movement rather than overcrowding the network to the server, thereby preventing the rate drop control of the server's unnecessary drop down / slow recovery.

In a multimedia session, the server sends multimedia data with an RTP header containing "RTP time stamp", "NTP time stamp" and sequence number information. The user who receives the RTP media packet can play the media in order using the RTP header information. In addition, the user uses the RTP header information to calculate packet loss and delay jitter to monitor the quality of service. Based on this quality of service condition, the user generates RTCP feedback information periodically and sends it to the server, helping the server to adjust the transmission rate in response to the ever-changing network conditions. When considering the handoff of the user, the user corrects the RTCP feedback information (cumulative number of packets lost, fraction lost and interarrival jitter) to hide the service quality degradation caused by the handoff, and the server gradually increases the rate control. It helps to maintain the existing rate without causing it. Thus, the present invention is referred to as fast recovery in terms of preventing the operation of slow recovery.

Hereinafter, the method of determining each service quality feedback information value in the existing RFC and how to correct the present invention to support mobility will be described.

In order to obtain a "cumulative number of packets lost", the user is the largest sequence number for a period of time t _i to the i-th transmitting RTCP packets from the start time t ₀ multimedia session When the multimedia session starts, the sequence number of the first RTP packet is defined as MIN. The difference between these two values is the number of packets expected to be received by the user until the i th packet is transmitted. Expressed as In addition, each time a user receives an RTP packet, the user counts and remembers the number of packets received. It is written as. Wow The difference in value That is, "cumulative number of packets lost".

= -MIN + 1,

= - .

If the user makes a handoff h between times t _i and t _i-1 , the user can detect his handoff and calculate the number L ^h of packets lost during the handoff period in a similar manner as above. When the user starts the handoff, he remembers the sequence number MIN ^h of the immediately preceding RTP packet and actually counts the number R ^h of the RTP packets he received during the handoff period. In addition, the sequence number MAX ^h of the first RTP packet received after the handoff is recorded, and the number of lost packets L ^h during the handoff period is calculated using Equation 1 using these three values.

L ^h = MAX ^h -MIN ^h -1 + R ^h .

When generating the RTCP information after the handoff, the user must add the number of packets lost L ^h during the handoff period to the number of packets lost from the start of the session to the present. By subtracting from the value "cumulative number of packets lost" can be corrected as shown in equation (compensated cumulative number of packets lost). As a result, the user hides the packet loss caused by his handoff to the server, and the server detects only the degradation of the service quality caused by the network congestion and improves the service quality by using the network congestion control method accordingly. Can be.

-L ^h .

"fraction lost" means the packet loss rate that occurred after sending the previous RTCP packet after sending the previous RTCP packet. In other words, the number of packets lost after sending the previous (i-1) RTCP packet and sending the current (i-th) RTCP packet. The number of packets expected to receive during the same period of It is defined as Equation 3 as a ratio to.

At this time, ego, to be.

Equation 3 also includes the number of packets lost during the handoff of the user. Equation 1 is subtracted from Equation 3 to correct the "fraction lost" value.

"interarrival jitter" means the average deviation of D, which is the difference in time it takes the user to receive the packet compared to the time it takes for the server to send one packet and the next. D is obtained by subtracting the time difference between time S _k and S _j from which each packet is transmitted from the time difference between time R _k when the k th RTP packet is received and the time R _{j when} the j th RTP packet is received.

D (k, j) = (R _j -R _k )-(S _j -S _k ) = (R _j -S _j )-(R _k -S _k ),

Using this D value and the previous "interarrival jitter" value, J _pre , the current "interarrival jitter" value, J, is obtained as in Equation 6.

The "interarrival jitter" value thus obtained does not need to be corrected. This means that sudden packet loss caused by handoff affects the "interarrival jitter" value based on Equations 5 and 6. Because it is not. Figure 1 shows the packet transmission pattern for each case when network congestion occurs, when the user moves in an ideal network condition.

1 is a diagram illustrating a form in which a packet is delivered according to a network state.

First, Fig. 1A shows a normal network condition, in which case the end-to-end delay of the packet is constant. Since the transmission interval s _i of the packet and the reception interval d _i of the packet are the same, D = 0, and therefore, the J value of Equation 5 does not change. 1B shows that the end-to-end delay of each packet gradually increases when network congestion occurs. In other words, the end-to-end delay of the current packet is larger than that of the previous packet, which results in an increase of J with an increase of D. Figure 1c shows the delivery form of a packet during handoff. As can be seen, the handoff does not affect the end-to-end delay of the packet during the period, but only results in the loss of packets delivered during the period, so that the D value between the packet just before the handoff occurs and the packet immediately after the handoff occurs It can be calculated by the difference between the transmission interval and the reception interval, and it can be seen that the calculation result is the same as the previous D, so that J does not change. This shows that J, the "interarrival jitter" value, is not affected by the handoff. 1B and 1C do not affect each other, so J, an "interarrival jitter", is changed only by network congestion. Each quality of service field value included in the RTCP feedback information can be corrected according to the above-described method. In the case of "Cumulative number of packets lost" and "fraction lost", the packet is corrected by ignoring packets lost during the handoff period based on Equations (2) and (4). On the other hand, the "interarrival jitter" is not affected by the handoff, so no correction is necessary.

FIG. 2 is a block diagram illustrating communication between a video server and a mobile host on which a service quality control method for guaranteeing user mobility is implemented in the Internet multimedia communication according to the present invention. The video server 10, the mobile host 12, and the forest agent are shown in FIG. (Foreign Agent) It consists of daemons. The mobile host 12 has a fast recovery module 14.

Referring to FIG. 3, the present invention is as follows.

3 is a flowchart illustrating a step-by-step embodiment of a service quality control method for guaranteeing user mobility in internet multimedia communication according to the present invention.

First, a user corresponding to a mobile host detects his or her handoff in a mobile IP network provided in an internet multimedia communication system (step 10).

While performing the detected handoff procedure, the mobile host collects information related to the quality of service it receives (step 12).

The information collected during the handoff period is used to calculate the number of packets lost during the handoff (step 14).

The number of lost packets during the handoff period is used to correct the " cumulative number of packets lost " and " fraction lost " values of the RTCP quality of service fields related to packet loss (step 16).

In order to calibrate the RTCP quality of service information value, the user should be able to count the number of packets lost during his handoff period. For this purpose, the user should be able to detect the start and end time of his handoff. In the present invention, it is assumed that the mobile network is a mobile IP network proposed by the Request For Comments (RFC) standard in order to guarantee the mobility of users in the Internet by the Internet Engineering Task Force (IETF). Mobile IP networks send a Mobile IP Registration Request (RREQ) message when the mobile host initiates a handoff, and upon successful handoff, the mobile host sends a Mobile IP Registration Reply (RREP) message. Will receive. Therefore, by capturing the packet, the handoff can be detected at the start and end of the handoff. It can be implemented using the IPtables module within the Netfilter framework supported by Linux kernel 2.4. This module can also be used to monitor the number of RTP packets a user receives during the handoff period. To detect handoffs, the RREQ and RREP packets mentioned above are captured using a specially assigned User Data Protocol (UDP) port number 434. Similarly, RTP packets are monitored using port numbers. In the case of RTCP packets, the quality of service field is corrected by capturing the port number specified for the RTCP session and the value of the RTCP packet type of the RTCP header using the receiver report type.

By capturing the RTP packet and the mobile IP registration request / response message, the fast recovery module located at the user records information regarding the quality of service (MIN ^h , MAX ^h , R ^h ) during handoff. These values are the information maintained by the existing video user module. Correct it by reflecting when making RTCP feedback information.

As described above, when the present invention is applied to TFRC, which is a transmission rate control scheme for multimedia, the present invention solves the degradation of service quality caused by network overload in fixed network, Deterioration of service quality due to network overcrowding can be prevented to reduce unnecessary transmission rate. First, it has the advantage of efficiently using the resources of the actual network. When the transmission rate is controlled using the conventional method, the mobility of the user is also gradually increased, which is the same as the network overload control, so that the actual available network resources are not properly utilized. The present invention does not inform the server of the service quality degradation such as packet loss caused by the mobility of the user, so that the server purely considers the degradation of the service quality caused by the network overcrowding, and thus the available network resources. Maintain more accurate information about Secondly, the present invention makes it possible to provide a quality of service suitable for fast recovery, which is a feature of the mobility of users compared to network overcrowding. When a user moves from one network to another network, a large amount of packet loss occurs temporarily during the moving process, but the service is provided at the original transmission rate from the moment the moving procedure is completed. The present invention has an advantage of providing better service quality by transmitting data at a data rate suitable for a user's capacity after moving by hiding information on packet loss occurring during the movement.

1 is a diagram illustrating a form in which a packet is delivered according to a network state;

FIG. 2 is a block diagram illustrating communication between a video server and a mobile host in which a service quality control method for guaranteeing user mobility is implemented in internet multimedia communication according to the present invention.

Figure 3 is a flow chart illustrating an embodiment of a service quality control method for ensuring user mobility in Internet multimedia communication according to the present invention step by step.

Claims (7)

In the Internet multimedia communication system having a handoff function, a user hides information about packet loss caused by his or her handoff, and reports the information on the quality of service to the server to the server, A first step of detecting, by the user corresponding to the mobile host, his or her handoff in a mobile IP network provided in the internet multimedia communication system; A second step of the mobile host collecting information related to a quality of service received by the mobile host while performing the sensed handoff procedure; A third step of calculating the number of packets lost during the handoff using the collected information during the handoff period; And And a fourth step of correcting the RTCP quality of service field and the RTCP feedback information related to packet loss by using the number of lost packets during the handoff period. The method of claim 1, wherein the handoff detection of the first step is performed based on a mobile IP handoff procedure performed by sending and receiving a mobile IP registration request and a mobile IP registration response message through a specific UDP port. A method of controlling a quality of service for guaranteeing user mobility in internet multimedia communication by monitoring a received message by a user detecting his or her handoff. The method of claim 1, wherein the second step, When receiving the mobile IP registration request message, the sequence number of the immediately received RTP data packet, MIN ^h, is stored, the RTP data packet actually received during the handoff h is counted, and the RTP received immediately after the R ^h handoff. A thirty-first step of recording a sequence number of the data packet, MAX ^h ; And And a thirty-second step of calculating the number of packets lost during the actual handoff period through the three values of step 31. 4. The method of claim 3, wherein the number L ^h of packets lost during the actual handoff period of step 32 is calculated by the following equation. [Equation] L ^h = MAX ^h -MIN ^h -1. 4. The method of claim 1, wherein the fourth step includes the number of packets lost L ^h during the handoff period and the number of packets lost from the session start (t ₀ ) to the present (t _i ). A quality of service control method for guaranteeing user mobility in Internet multimedia communication that corrects the RTCP quality of service field by ss. The method of claim 1, wherein the fourth step corrects the RTCP feedback information by using the number of packets lost L _h during the handoff period as in the following equation. [Equation] The method of claim 1, wherein the RTCP quality of service field is "cumulative number of packets lost" and the RTCP feedback information is "fraction lost". Quality control method.