KR100631516B1 - Streaming system and adaptive band allocation method - Google Patents

Abstract

The present invention relates to a streaming system and an adaptive band allocation method. In the related art, a streaming system dynamically adjusts a transmission bit rate to match an average bit rate of an audio / video encoder to prevent overflow and underflow of a receiving streaming buffer. . In consideration of such a necessity, the present invention obtains buffer level and monitoring information according to a report message of a client, allocates resources by adjusting timer frequency and bandwidth according to the client's status information, and generates multimedia data in the form of an RTP packet and transmits it to the client. A streaming server; Based on the client-side buffer level, it consists of a client that sets the data bandwidth according to the streaming server's resource allocation policy, extracts the monitoring information for the incoming media stream, writes the buffer level and the monitoring information as a report message, and delivers it to the streaming server. By dynamically adjusting the server side bit rate, it is possible to prevent the client side buffer from abnormal condition and to prevent packet loss due to buffer overflow / underflow.

Description

Streaming system and adaptive band allocation method {STREAMING SYSTEM AND ADAPTIVE BANDWIDTH ALLOCATION METHOD}

1 is a block diagram showing the configuration of a conventional streaming system.

2 is a block diagram showing a server configuration of a streaming system according to an embodiment of the present invention.

3 is a block diagram showing a client configuration of a streaming system according to an embodiment of the present invention.

4 is a flowchart illustrating an adaptive band allocation method of a streaming system according to an exemplary embodiment of the present invention.

5 is an exemplary diagram illustrating an overflow of a buffer, an underflow of a buffer, and an empty buffer caused by packet loss when flow control is not performed in a client of a streaming system.

** Description of the symbols for the main parts of the drawings **

22: Streaming Engine 25: Quality of Service Manager

36: Quality of Service Client 37: Quality of Service Monitor

The present invention relates to a streaming system and an adaptive band allocation method. In particular, the present invention relates to a streaming system and an adaptive band allocation method capable of improving the quality of service by adaptively allocating network resources based on a buffer level and a packet error rate on a client side. It is about.

The home network system includes a wireless network module to provide data-based services such as file transmission and reception and Internet access, as well as streaming multimedia contents. Unlike the operation of reading and playing a video file from a storage medium, the streaming service outputs a video by simultaneously downloading and playing a voice and video from a server.

The conventional streaming system guarantees the quality of a streaming service in a wireless environment by gradually adjusting the streaming bit rate by using the state information of the wireless network based on the statistical information of the received packet. The method of adjusting the streaming bit rate is used when a server provides a streaming service using a file encoded in an audio and video format that allows a variable bit rate, and when streaming a content encoded by encoding a content at a real-time variable bit rate. do.

1 is a block diagram showing the configuration of a conventional streaming system, as shown therein, a streaming server 11 for streaming data through a designated socket connection upon request from a client side; Clients 12 and 13 receive the streamed data, store it in the streaming buffer, and deliver the receiving side statistical information to the streaming server.

The streaming server 11 stores multimedia content in a storage medium of the hard disk and streams data through a designated socket connection when a content request is received from the client 12 or 13. At this time, the streaming server 11 analyzes the header of the stored stream, checks the format and data bandwidth of the stream to be transmitted, determines the data transmission rate, and transmits data through the socket.

The streaming server 11 transmits the sending side statistical information (time stamp, cumulative number of packets) of the media session connected with the client 12, 13 to the client 12, 13, and the client 12, 13 receives Side statistics information (cumulative lost packet count, packet jitter) is transmitted to the streaming server 11. The streaming system defines the necessary monitoring information and sends and receives the monitoring information between the server and the client to determine the status of the counterpart for the streaming session.

The streaming system succeeds in controlling the flow between the server and the client and tries to improve the quality of service by transmitting real-time data on the IP network. In other words, the streaming system dynamically adjusts the transmission bit rate to meet the average bit rate required by the audio / video encoder to prevent overflow and underflow of the receiving streaming buffer.

In addition, the conventional streaming method wirelessly converts multimedia data encoded with a fixed bit rate encoder by using a file encoded in an audio or video format allowing a variable bit rate on a server side or by mounting a real time variable bit rate encoder. There is a need for a method of improving the quality of service when transmitting on a network.

Accordingly, the present invention has been made in view of the above necessity, and is capable of dynamically adapting and allocating network resources based on the buffer level and packet error rate on the client side and improving the quality of service by adjusting the server side data rate. Its purpose is to provide a streaming system and an adaptive band allocation method.

In order to achieve the above object, the present invention obtains a buffer level and monitoring information by a report message of a client, allocates resources by adjusting a timer frequency and a bandwidth according to the client's state information, and generates multimedia data in the form of an RTP packet. Streaming server for transmitting to the client; According to the resource allocation policy of the streaming server, the data band is set, and the monitoring information of the received media stream is extracted, and the buffer level and the monitoring information are written as a report message.

In addition, setting the bit rate to the required minimum bit rate, starting the streaming service, and starting to receive the RTCP monitoring data; Extracting a packet error rate and a buffer level from the RTCP monitoring data; Determining whether the buffer level exceeds an upper limit threshold and, if exceeding, decreasing the timer frequency by a predetermined percentage; otherwise, determining whether the packet error rate is zero; Determining whether the buffer level is less than the lower limit threshold when the packet error rate is zero, and increasing the timer frequency by a predetermined percentage when the packet error rate is less than, and decreasing and adjusting the bit rate by a predetermined bandwidth when the bit rate is different from the initial bit rate; When the packet error rate is not zero, it is determined whether or not the buffer level is less than the lower limit threshold.

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

2 is a block diagram showing a server configuration of a streaming system according to an exemplary embodiment of the present invention, which includes a data storage unit 21 for storing multimedia data; A streaming engine 22 which receives the multimedia data and generates a RTP packet, and outputs the multimedia data to the client through the socket interface 24 in real time by controlling the output time of the timer 23; A quality of service manager 25 for obtaining buffer level and monitoring information through an RR message generated by the quality of service monitor of the client and determining a policy regarding required resource allocation using server-side / client-side state information; It consists of a signaling interface 26 that sets the data band required by the media stream through the parameterized quality of service support function supported by the media control layer of the wireless module 27 according to the resource allocation policy of the quality of service manager 25. do.

The data storage unit 21 stores multimedia data, and the streaming engine 22 receives the multimedia data from the data storage unit 21 and generates the RTP packet in real time by controlling the output time of the RTP packet of the timer. Output via interface 24.

The socket interface 24 delivers the RTP packet to the radio module 27 through the network layer, and the radio module 27 provides a media control layer and converts the digital data of the RTP packet into radio frequency signals to transmit and receive.

The quality of service manager 25 uses the server side and client side state information to determine the necessary policy regarding resource allocation, and the state information of jitter, delay, throughput, and packet error rate through an RTCP RR (Receiver Report) message on the client side. Get the buffer level and monitoring information according to the application through the RR message generated by the quality of service monitor.

The signaling interface 26 sets the data band required by the media stream through a parameterized quality of service support function supported by the media control layer of the wireless module 27 according to the resource allocation policy of the quality of service manager 25.

3 is a block diagram illustrating a client configuration of a streaming system according to an exemplary embodiment of the present invention. As shown in FIG. 3, a socket interface for transmitting an RTP packet of a wireless module 31 to a streaming buffer 33 is provided through a network layer. 32); A signaling interface 35 for setting a data band required by the media stream of the socket interface 32 according to the resource allocation policy of the quality of service manager of the streaming server; A quality of service client 36 that controls the wireless module 31 through the signaling interface 35 in accordance with the band allocation policy determined by the quality of service manager of the streaming server; There is one instance for each media session, and it extracts monitoring information about RTP packet and media stream received through SR of RTCP, and sends buffer level and monitoring information to quality of service manager of streaming server through RTCP RR message. It consists of a quality of service monitor 37 to deliver.

The radio module 31 provides a media control layer and converts radio frequency signals into digital data to receive RTP packets, and the socket interface 32 transmits the RTP packets of the radio module 31 through the network layer to the streaming buffer 33. To pass).

The streaming buffer 33 rearranges the order of the RTP packets not sequentially received, removes the headers from the rearranged RTP packets in the order transmitted from the streaming server, and outputs the headers to the audio / video decoder 34. 34) decodes the RTP packet and outputs audio / video to the display unit.

The signaling interface 35 sets a data band required by the media stream through a parameterized quality of service support function supported by the media control layer of the wireless module 31 according to the resource allocation policy of the quality of service manager of the streaming server. The quality of service client 36 controls the wireless module 31 via the signaling interface 35 in accordance with the band allocation policy determined by the quality of service manager of the streaming server.

The quality of service monitor 37 has one instance for each media session, extracts the monitoring information for the media stream received through the RTP packet and the sender report (SR) of the RTCP, and uses the RTCP RR message to Deliver to Quality of Service Manager.

4 is an operation flowchart of an adaptive band allocation method of a streaming system according to an exemplary embodiment of the present invention. As shown in FIG. 4, a bit rate is set to a required minimum bit rate, a streaming service is started, and RTCP monitoring data is received. Steps; Extracting a packet error rate and a buffer level from the RTCP monitoring data; Determining whether the buffer level exceeds an upper limit threshold, if exceeding, decreasing the timer frequency by 0.1 percent, and if not, determining whether the packet error rate is zero; Determining whether the buffer level is less than the lower threshold when the packet error rate is zero, and increasing the timer frequency by 0.1 percent when the packet error rate is less than, and reducing and adjusting the bit rate by a predetermined bandwidth when the bit rate is different from the initial bit rate. If the packet error rate is not zero, it is determined whether or not the buffer level is lower than the lower limit threshold.

The quality of service manager uses parameters such as jitter, delay, throughput, packet error rate, and client-side buffer level as criteria for adjusting bandwidth allocation and server-side bit rate. Client-side buffer levels are the levels of network layer buffers (socket buffers) and play-out buffers involved in streaming. The quality of service manager interprets the RR / SR message provided by RTP / RTCP to get the status information of parameters except the buffer level, and obtains the buffer level monitoring information by defining application-defined packets in RTP / RTCP.

4 bytes 4 bytes 4 bytes 4 bytes 4 bytes V / P / Subtype / PT = 204 / Length SSRC "APP1" Sock_buffer_level Playout_buffer_level

For example, the application-defined packet is 4 bytes for version, padding, payload type, 4 bytes for SSRC (Synchronization Source), 4 bytes for "APP1", 4 bytes for socket buffer level, and play out buffer level as shown in Table 1. Consists of 4 bytes.

The quality of service manager sets the bit rate to the minimum bit rate required, starts the streaming service, and starts receiving RTCP monitoring data.

The quality of service manager adjusts the timer frequency and the bit rate by referring to the buffer level and the packet error rate until the service is stopped. The quality of service manager extracts the packet error rate and the buffer level from the RTCP monitoring data.

The quality of service manager determines whether the buffer level exceeds the upper limit threshold, and if it exceeds, decreases the timer frequency by 0.1 percent to delay the RTP packet output time, and if not, determines whether the packet error rate is zero.

Scenarios that can occur during data streaming include: if the packet error rate is zero (the current wireless state is good), if a packet error occurs occasionally (the current wireless state is streaming, but the wireless state is not good), It is divided into three cases that occur frequently (in a wireless state where streaming is not possible).

The quality of service manager determines whether the packet error rate is zero or not, classifies cases that may occur during data streaming, and adjusts the timer frequency and bit rate according to each situation.

If the packet error rate is zero, the quality of service manager determines whether the buffer level is less than the lower limit threshold, and if it is less than, increases the timer frequency, and if it is less than, compares the bit rate with the initial bit rate and reduces and adjusts the bit rate by a predetermined bandwidth.

If the packet error rate is zero, the streaming service is sufficient with only the initially allocated bit rate during streaming and does not require a separate bandwidth allocation policy, but only corrects the bit rate value changed by the service quality manager in the previous time. That is, the quality of service manager increases the timer frequency when the buffer level is lower than the lower limit threshold, and decreases the bit rate by a predetermined bandwidth when the buffer level is higher than the lower limit threshold and the bit rate and the initial bit rate are different.

If the packet error rate is not zero, the quality of service manager determines whether the buffer level is less than the lower limit threshold, and if it is less, increases and adjusts the bit rate by a predetermined bandwidth. If packet errors sometimes exist, the quality of service manager increases the bit rate by a predetermined bandwidth (e.g., 5 percent of the bit rate) at the initial allocated bit rate, taking into account retransmission.

If a packet error always exists (not shown), the service quality manager cannot change the bandwidth allocation policy to guarantee the required quality of service, so the user is notified of the streaming service and stops the service.

5 is an exemplary diagram illustrating an overflow of a buffer, an underflow of a buffer, and an empty buffer caused by packet loss when flow control is not performed in a client of a streaming system.

Although the packet error rate is zero in the streaming system, if the flow control is not performed in the unidirectional streaming, overflow (point A in FIG. 5A) to underflow (point B in FIG. 5B) occurs in the client side buffer. The client-side buffer fault condition is identical to the situation where a packet loss occurred to an end user. If a buffer overflow occurs, the streaming server transmits data faster than the bit rate required by the client's decoder. If a buffer underflow occurs, the streaming server's average bit rate is higher than the bit rate required by the client's decoder. It is slow.

The quality of service manager varies the timer frequency to dynamically adjust the average transmission bit rate of the server-side streaming engine using the buffer level information from the quality of service monitor of the client to prevent the occurrence of a buffer abnormal condition. That is, the quality of service manager decreases the timer frequency by a predetermined ratio of the current value when the buffer level (socket buffer level + play out buffer level) is larger than the upper limit threshold, and increases by a predetermined ratio of the current value by less than the lower limit threshold.

When there is a packet loss, as shown in FIG. 5C, the buffer level is significantly changed at the point C due to the packet loss at the point C, but the buffer underflow does not occur, and at the point D, the packet loss is large. Buffer emptying occurred. If the buffer continues to empty, the quality of service manager should stop the streaming service. That is, the quality of service manager stops the streaming service when the time when the buffer level is zero continues for a predetermined time.

 As described above in detail, the present invention implements an adaptive band allocation policy based on monitoring data for a session unit in fixed bit rate traffic, and adjusts the band allocation and the timer frequency in response to a change in radio conditions, The quality of service is improved by satisfying the required data reception rate.

In addition, by dynamically adjusting the server-side bit rate based on the client-side buffer level, there is an effect of preventing packet loss due to buffer overflow / underflow by preventing an abnormal state of the client-side buffer.

Claims (6)

A streaming server obtaining buffer level and monitoring information by a client report message, allocating resources by adjusting a timer frequency and a bandwidth according to client status information, and generating multimedia data in the form of an RTP packet to the client; A streaming system comprising a client that sets a data bandwidth according to a resource allocation policy of a streaming server, extracts monitoring information on a received media stream, writes a buffer level and monitoring information as a report message, and delivers the result to a streaming server. The streaming server of claim 1, further comprising: a streaming engine generating multimedia data in the form of an RTP packet and outputting the multimedia data to a client through a socket interface in real time by controlling an output time of a timer; A quality of service manager for obtaining a buffer level and monitoring information through an RR message generated by a quality of service monitor of a client and adjusting a timer frequency and a bandwidth using server-side / client-side state information to determine a policy regarding resource allocation required; And a signaling interface configured to set a data band required by the media stream through a parameterized quality of service support function supported by the media control layer of the wireless module according to a resource allocation policy of the quality of service manager. 2. The apparatus of claim 1, wherein the client further comprises: a signaling interface configured to set a data band required by the media stream of the socket interface according to a resource allocation policy of the quality of service manager of the streaming server; A quality of service client for controlling the wireless module through the signaling interface according to the bandwidth allocation policy determined by the quality of service manager of the streaming server; There is one instance for each media session and extracts monitoring information about media streams received through RTP packets and RTCP report messages from the server, and buffer level and monitoring information to the quality of service manager of the streaming server via RTCP report messages. Streaming system, characterized in that consisting of a quality of service monitor for delivering. Setting the bit rate to the required minimum bit rate, starting a streaming service, and starting to receive RTCP monitoring data; Extracting a packet error rate and a buffer level from the RTCP monitoring data; Determining whether the buffer level exceeds an upper limit threshold and, if exceeding, decreasing the timer frequency by a predetermined percentage; otherwise, determining whether the packet error rate is zero; Determining whether the buffer level is less than the lower limit threshold when the packet error rate is zero, and increasing the timer frequency by a predetermined percentage when the packet error rate is less than, and decreasing and adjusting the bit rate by a predetermined bandwidth when the bit rate is different from the initial bit rate; And if the packet error rate is not zero, determining whether the buffer level is less than the lower limit threshold, and if the packet error rate is less than zero, increasing and adjusting the bit rate by a predetermined bandwidth. 5. The method of claim 4, wherein the buffer level is a level calculated by adding a socket buffer level and a play out buffer level. 5. The method of claim 4, further comprising: notifying the user that the streaming service is unavailable and stopping the service if the packet error rate is not zero and determines that the time when the buffer level is zero continues for a predetermined time. Adaptive band allocation method in system.

Images