WO2012053834A2 - Dispositif et procédé d'émission/réception de service de diffusion en mode continu - Google Patents

Dispositif et procédé d'émission/réception de service de diffusion en mode continu Download PDF

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Publication number
WO2012053834A2
WO2012053834A2 PCT/KR2011/007827 KR2011007827W WO2012053834A2 WO 2012053834 A2 WO2012053834 A2 WO 2012053834A2 KR 2011007827 W KR2011007827 W KR 2011007827W WO 2012053834 A2 WO2012053834 A2 WO 2012053834A2
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WIPO (PCT)
Prior art keywords
tcp packet
time
timestamp
jitter
arrival
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PCT/KR2011/007827
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English (en)
Korean (ko)
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WO2012053834A3 (fr
Inventor
정순흥
서광덕
이진영
배성준
심상우
탕쯔엉공
박상택
류원
Original Assignee
한국전자통신연구원
연세대학교 산학협력단
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Priority claimed from KR1020110030790A external-priority patent/KR101180540B1/ko
Application filed by 한국전자통신연구원, 연세대학교 산학협력단 filed Critical 한국전자통신연구원
Priority to US13/880,941 priority Critical patent/US20130282871A1/en
Publication of WO2012053834A2 publication Critical patent/WO2012053834A2/fr
Publication of WO2012053834A3 publication Critical patent/WO2012053834A3/fr

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/4302—Content synchronisation processes, e.g. decoder synchronisation
    • H04N21/4305—Synchronising client clock from received content stream, e.g. locking decoder clock with encoder clock, extraction of the PCR packets
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]

Definitions

  • the present invention relates to a streaming service transmission / reception system, and more particularly, to an apparatus and method for progressive streaming service transmission / reception using de-jitter buffering time estimation.
  • IP Internet Protocol
  • a multimedia payload is packetized into a Real-time Transport Protocol (RTP), and then an RTP packet is loaded on a user datagram protocol (UDP) and transmitted.
  • RTP Real-time Transport Protocol
  • UDP user datagram protocol
  • UDP is suitable for real-time multimedia transmission but does not provide the ability to improve the quality of service (QoS) by retransmission of packets that are lost during transmission.
  • QoS quality of service
  • Digital video such as H.264
  • Digital video is compressed by using prediction based coding between frames.
  • the error due to the lost packet is followed until the blocked video is blocked by a subsequent I-frame received after the lost packet. Propagated through the frames.
  • RTP / UDP-based real-time multimedia streaming is suitable for a real-time service requiring low delay, but has a disadvantage in terms of guaranteeing QoS.
  • the download method If the download method is used, the client system downloads the entire video file and then starts playback. Therefore, the user of the client system can enjoy high quality service.
  • the download method has the same problems as 1) and 2) below.
  • the user can watch the video after a long download time.
  • the client side must prepare a storage space to download and save the entire video file in advance.
  • Progressive streaming services are similar to downloads in that they use the transmission control protocol (TCP) as the transport protocol.
  • TCP transmission control protocol
  • the progressive streaming service is similar to the real-time streaming in that it plays the media data already received in real time while the file including the media data is downloaded.
  • progressive streaming service can be regarded as a technology that utilizes only the advantages of both download and streaming technologies.
  • MPEG transport stream is a transmission protocol standard standard for multiplexing compressed data of audio and video and delivering multiplexed compressed data through a broadcasting network such as DVB, ISDB, ATSC, DMB, and the like.
  • a streaming service for transmitting MPEG TS through a wired or wireless IP network by using HTTP and TCP protocols may be provided.
  • a streaming service transmitting / receiving apparatus and method for transmitting a TCP packet including information necessary for calculating arrival interval jitter may be provided.
  • One embodiment of the present invention can provide a streaming service transmission and reception apparatus and method for calculating the arrival interval jitter based on the arrival time and the expected arrival time of the TCP packet.
  • a streaming service transmission method is provided, which is the expected arrival time of.
  • the streaming service transmission method may further include checking whether there is a TS packet including the PCR information in a TS header among the one or more TS packets loaded in the payload.
  • the timestamp may represent a time at which the first byte of the payload arrives at the terminal as a system time clock count value using the system clock frequency of the streaming server.
  • the time stamp may include a base clock and an extended clock, and the base clock may use a smaller frequency unit than the extended clock.
  • the length of the bits of the timestamp may be determined based on the precision of the arrival time delay variation required by the terminal.
  • Writing the timestamp in the TCP packet may store the timestamp in an options field in the header of the TCP packet.
  • the TS packet load unit for loading one or more TS packets in the payload of the TCP packet, setting the first PCR information in the one or more TS packets in the payload to the PCR of the TCP packet
  • a PCR setting unit for calculating a timestamp of the TCP packet based on the PCR, a timestamp recording unit for recording the timestamp in the TCP packet, and a transmission unit for transmitting the TCP packet to the terminal;
  • a time stamp is provided with a streaming service transmitting apparatus, wherein the TCP packet is an expected arrival time of the TCP packet.
  • the first timestamp is the reception time of the first TCP packet expected by the server
  • the second clock count is a system time clock at the moment a second TCP packet received before the TCP packet is received, and the second time
  • the stamp is provided with a streaming service receiving method, wherein the server is the time of receipt of the second TCP packet expected by the server extracted from the second TCP packet.
  • the operation of calculating a de-jitter buffering time based on the first clock count, the first timestamp, the second clock count, and the second timestamp may include: generating a first jitter based on the first clock count and the first timestamp; Calculating a first network jitter experienced by a TCP packet, a first arrival between the second TCP packet and the first TCP packet based on the first network jitter and the second network jitter experienced by the second TCP packet Calculating a time interval and calculating the de-jitter buffering time based on the first network jitter and the first arrival time interval.
  • the calculating of the de-jitter buffering time based on the first network jitter and the first arrival time interval may include: determining the de-jitter buffering time based on the second arrival interval jitter and the first arrival time interval of the second TCP packet. Calculating a first arrival interval jitter of one TCP packet, wherein the arrival interval jitter is an arrival interval delay variation between successive arriving packets, a second arrival interval variance of the second TCP packet, the first arrival time Calculating a first arrival interval variance of the first TCP packet based on an interval and the first arrival time interval, wherein the arrival time variance is a variance for the arrival time interval and a first first arrival interval jitter and the Calculating the de-jitter buffering time based on a first arrival interval variance.
  • the first timestamp may be in an options field in the header of the first TCP packet.
  • the streaming service receiving method may further include determining a final de-jitter buffering time among the de-jitter buffering time and the analyzing time, wherein the analyzing time is a time sufficient to receive and analyze the received TCP packets. Can be.
  • the streaming service receiving method may further include setting the de-jitter buffering time to a buffering time of a transport stream system target decoder of the streaming client.
  • a receiver for receiving a first TCP packet, a clock count measuring unit for measuring a first clock count of the system time clock at the moment the first TCP packet is received, from the first TCP packet
  • a timestamp extractor extracting a first timestamp of the first TCP packet and a de-jitter buffering time calculated based on the first clock count, first timestamp, second clock count, and second timestamp
  • a jitter buffering time calculation unit wherein the first timestamp is a reception time of the first TCP packet expected by the server, and the second clock count is a value of receiving a second TCP packet received before the TCP packet;
  • a second system time clock wherein the second time stamp is a reception time of the second TCP packet expected by the server extracted from the second TCP packet.
  • a streaming service transmitting / receiving apparatus and method for calculating arrival interval jitter based on arrival time and expected arrival time of a TCP packet are provided.
  • FIG 1 illustrates an overview of an HTTP-based progressive streaming service according to an example of the present invention.
  • FIG. 2 shows an overview of an MPEG TS based progressive streaming service according to an embodiment of the present invention.
  • FIG. 3 illustrates a structure of an MPEG TS packet according to an embodiment of the present invention.
  • FIG. 4 illustrates a standard T-STD buffer model according to an example of the present invention.
  • FIG 5 illustrates a change in packet arrival delay time in an IP network according to an embodiment of the present invention.
  • FIG. 6 illustrates a TCP payload format composed of n MPEG TS packets according to an embodiment of the present invention.
  • FIG. 7 illustrates a structure of a TCP packet including TS packets including PCR information in a header according to an embodiment of the present invention in a TCP payload.
  • FIG. 8 illustrates a timing model for estimating the time at which a TCP packet arrives at a receiver according to an example of the present invention.
  • FIG. 9 is a flowchart illustrating a method of operating a transmitting side according to an embodiment of the present invention.
  • FIG. 10 illustrates a basic structure of a header 310 of a TCP packet and an extension field that may be selectively used according to an embodiment of the present invention.
  • FIG. 11 is a structural diagram of a transmitting side 1100 according to an embodiment of the present invention.
  • FIG. 12 is a flowchart illustrating a method of operating a receiving side according to an embodiment of the present invention.
  • FIG. 13 illustrates a T-STD buffer model including de-jitter buffering according to an embodiment of the present invention.
  • FIG. 14 is a structural diagram of a receiving side 1400 according to an example of the present invention.
  • 15 illustrates a delay of media playback time due to de-jitter buffering configured according to an embodiment of the present invention.
  • FIG 1 illustrates an overview of an HTTP-based progressive streaming service according to an example of the present invention.
  • the terminal (ie, client) 110 transmits a "GET Request" 130 of the HTTP protocol to inform the server 120 of the service request for the desired content file.
  • the server 120 transmits the content file to the terminal 110.
  • the server 120 uses TCP as a transport protocol for transmitting the content file to guarantee QoS.
  • the content file arrives at the terminal 110 by TCP.
  • the terminal 110 buffers some data of an initially arrived content file.
  • the terminal 110 displays the data in the TCP packet arriving in real time from when the TCP packet arriving at the terminal 110 fills the buffer by a predetermined amount in order to reduce the service delay time, which is a problem of the conventional download scheme. . That is, the terminal 110 parses some data of the buffered content file to play out video and audio.
  • the HTTP-based progressive streaming service is differentiated from the download method in which the content file is played only after all data of the content file arrives at the terminal 110.
  • the progressive streaming service based on HTTP can reduce service delay time, which is a problem of the conventional download method, and can guarantee QoS.
  • Content files transmitted from existing progressive streaming services are QuickTime (.mov), MPEG-4 (.mp4) and Windows Media Video (Windows Media Video) to store compressed video and audio as a single file. (.wmv), Flash Video (.flv), and 3GPP (.3gp) file formats.
  • FIG. 2 shows an overview of an MPEG TS based progressive streaming service according to an embodiment of the present invention.
  • FIG. 2 shows an overview of a service for providing multimedia content produced by the MPEG TS standard by HTTP progressive streaming.
  • the HTTP progressive streaming server can produce (i.e. encode) video / audio compressed data in MPEG TS standards compressed with various multimedia encoders for use as MPEG TS broadcast content.
  • the produced MPEG TS content may be serviced through HTTP progressive streaming.
  • the HTTP progressive streaming server may utilize the MPEG TS content obtained by converting a content file produced by an existing file format standard into MPEG TS for the HTTP progressive streaming service.
  • a client using HTTP progressive streaming simply requests the desired MPEG TS content from the server through an HTTP GET request.
  • a program clock reference included in a header of an MPEG TS packet when MPEG TS content is transmitted to a client through an IP network (eg, a wired or wireless Internet) by a progressive streaming service, a program clock reference included in a header of an MPEG TS packet.
  • IP network eg, a wired or wireless Internet
  • a method and apparatus for effectively estimating network jitter by utilizing PCR clock information are disclosed.
  • Network jitter is the arrival time delay variation between TCP packets arriving at the receiving end.
  • Embodiments of the present invention may estimate inter-arrival jitter, which is a delay variation in arrival time between packets based on network jitter.
  • T-STD Transport Stream System Target Decoder
  • the de-jitter buffering of the terminal absorbs the difference in arrival time with the packets arriving later by not allowing the receiving side to start the playback process immediately when the first packet arrives at the receiving side.
  • the de-jitter buffering allows the receiving side to start playing after collecting the packets to some extent so that the original time interval between the packets can be maintained.
  • the length of de-jitter buffering is that after the first packet arrives at the receiver, before the receiver performs the full video and audio playback process, the first packet must wait in the file buffer. Decide when to do it.
  • the initial playout process immediately following the de-jitter buffering is delayed in proportion to the size of the buffering.
  • the latency for services experienced by the user increases.
  • Embodiments of the present invention are directed to the size of the appropriate de-jitter buffering time and the size of the de-jitter buffering required to absorb arrival interval jitter through estimation of arrival interval jitter of TCP packets in an MPEG TS based progressive streaming service.
  • a method and apparatus for determining a corresponding initial playout delay is disclosed.
  • the initial play delay time corresponds to the time after the first packet arrives at the receiver, and the first packet stays for de-jitter buffering in the TCP file buffer before the packet is delivered to the T-STD buffer model.
  • the initial playback delay time determines the initial playback time point for the received media (or content).
  • FIG. 3 illustrates a structure of an MPEG TS packet according to an embodiment of the present invention.
  • the MPEG TS packet 300 is a fixed length packet of size 188 bytes.
  • MPEG TS is a continuous stream of MPEG TS packets 300.
  • the structure of the header 310 and payload 320 of the MPEG TS packet 300 is shown.
  • the fields in the header 310 are shown hierarchically. The number below the field indicates the size (in bits) of the field.
  • the header 310 is basically 4 bytes in size (sync byte field to continuity counter field).
  • the size of the header 310 can be increased according to the presence or absence of an adaptation field and an optional field existing to deliver various extended information.
  • MPEG TS is a standard for effectively multiplexing and delivering video and audio compressed data transmitted from a digital broadcasting service.
  • the elementary stream of compressed video / audio is packetized elementary stream (PES) packetized.
  • PES packetized elementary stream
  • the PES packet is finely divided into fixed length TS packets of 188 bytes in size, and the divided TS packets are transmitted by properly multiplexing video and audio.
  • FIG. 4 illustrates a standard T-STD buffer model according to an example of the present invention.
  • MPEG TS is a standard developed for the purpose of digital broadcasting service. Therefore, TS packets are delivered to the receiver through a broadcasting network which is a circuit switched network in which channel quality is relatively stable. Therefore, the packet delay time experienced by MPEG TS packets in the transport channel is relatively short and constant. In addition, a timing buffer model for sequentially processing TS packets arriving at the receiver may also be effectively applied.
  • T-STD Transport Stream System Target Decoder
  • FIG. 4 a structure of a standard T-STD buffer model is disclosed in which multiplexed TS packets related to video, audio, and system information received by multiplexing MPEG MPEG are demultiplexed, and the demultiplexed streams are sequentially buffered.
  • MPEG TS aims to deliver multimedia data over circuit switched networks with relatively high bandwidth quality and low propagation delay, such as broadcast networks or Asynchronous Transfer Mode (ATM) networks.
  • ATM Asynchronous Transfer Mode
  • the delay variation in arrival time between packets which is a chronic problem of the Internet, may be estimated accurately.
  • the standard T-STD model of FIG. 4 designed for MPEG TS transmission through a broadcasting network is also required to be modified based on this estimation.
  • each of the packets When packets are transmitted through the wired / wireless Internet based on the IP network, the routing paths through which each of the packets may be different from each other, unlike when the packets are transmitted through the existing broadcasting network. In addition, since the amount of traffic to be processed and the processing time for each router through each of the packets are different from each other, the destination arrival times of each of the packets may be different from each other.
  • An embodiment of the present invention to be described below discloses a method for accurately estimating arrival delay time variation between TCP packets occurring when MPEG TS is delivered to a client via a progressive streaming service in a wired or wireless Internet environment.
  • embodiments of the present invention to be described below disclose a modified T-STD model based on the estimated arrival delay variation.
  • FIG 5 illustrates a change in packet arrival delay time in an IP network according to an embodiment of the present invention.
  • FIG. 5 shows an example of a dynamic variation in the arrival delay time that packets undergo when they are delivered over a general IP network.
  • packets of size h bytes are transmitted through the IP network.
  • the packets undergo a transmission delay of x1, x2 and x3 through the IP network and arrive at the receiver at t1, t2 and t3 time instants, respectively.
  • time x3 is larger than x1 and x2.
  • a variation of arrival delay time by b seconds occurs between t2 and t3.
  • the size of the de-jitter buffering is too small, the risk of buffer underflow increases, resulting in frequent refurbishing. Conversely, if the size of the de-jitter buffering is too large, the buffering time for the arrived data becomes long, resulting in latency of service resulting in poor service quality.
  • Inter-arrival jitter is a variation in arrival delay time between packets.
  • PCR may be present in an MPEG TS packet header carried in a TCP packet.
  • the computational and implementation complexity of the method and apparatus for estimating inter-arrival jitter by utilizing the PCR information recorded in the TS packet is low.
  • FIG. 6 illustrates a TCP payload format composed of n MPEG TS packets according to an embodiment of the present invention.
  • MPEG TS packets have a fixed size of 188 bytes. Therefore, several TS packets may be carried in the payload of one TCP packet. Considering that the size of the Ethernet maximum transfer unit (MTU) is 1500 bytes, theoretically, up to seven TS packets may be carried in the payload of TCP.
  • MTU Ethernet maximum transfer unit
  • a TCP payload format is shown that maps n TS packets into the payload of one TCP packet.
  • PCR is present in the header of the TS packet.
  • the PCR value is a value obtained by sampling the time at the system encoder at a system clock frequency (SCF) of 27 MHz.
  • the PCR clock can consist of a total of 42 bits.
  • the PCR clock may consist of a 33-bit PCR_base expressed in units of 90 KHz (ie, SCF / 300) and 9 bits of PCR_ext expressed in units of 27 MHz (ie, SCF).
  • PCR is typically transmitted periodically within 100 ms to maintain STC synchronization between the transmitter and receiver.
  • TS packet including PCR information in a header among TS packets constituting the payload format of the TCP packet.
  • FIG. 7 illustrates a structure of a TCP packet including TS packets including PCR information in a header according to an embodiment of the present invention in a TCP payload.
  • TCP (n) 710 and TCP (n + 1) 750 are shown, respectively.
  • TCP (n) 710 is an n th TCP packet containing TS packets including PCR information in a header in a TCP payload.
  • TCP (n + 1) 750 is the n + 1 th TCP packet which contains TS packets including PCR information in the header in the TCP payload.
  • TCP (n) and TCP (n + 1) are two TCP packets whose transmission order sequentially neighbors, while including at least one TS packet including PCR information in a header in the TCP payload.
  • a method of using PCR information included in the TCP payload to estimate the arrival time delay variation of the TCP packet arriving at the receiving side is described below.
  • FIG. 8 illustrates a timing model for estimating the time at which a TCP packet arrives at a receiver according to an example of the present invention.
  • TCP (n) which is the nth TCP packet, includes a plurality of consecutive TS packets in the payload, in a format as described above with reference to FIG. 7.
  • N is an integer of 0 or more.
  • the horizontal axis of FIG. 8 represents a theoretical time at which a TCP packet arrives at a receiving side and a corresponding TCP timestamp value.
  • the vertical axis represents the index value of each successive data byte sent to the receiver.
  • TCP (n) utilizes the PCR value of the first TS packet including PCR information as PCR (n), which is a PCR value representative of TCP (n).
  • n is the index of TCP. That is, the value of n in TCP (n).
  • PCR (n) is the index of the first PCR in TCP (n).
  • i n is the byte index of PCR (n), and i n + 1 is the byte index of PCR (n + 1).
  • the time t (i) at which the i-th data byte arrives at the receiver may be estimated according to Equation 1 below.
  • R (n) is the data rate between the PCR (n) and PCR (n + 1) clocks (ie, the transport rate of the transport stream).
  • R (n) may be calculated according to Equation 2 below.
  • the time t (n + 1) at which the first byte of the payload of TCP (n + 1) arrives at the receiver may be calculated according to Equation 3 below.
  • I represents the size of all the data transmitted in bytes by being located ahead of the byte including the last bit indicating the information of PCR (n + 1) among all data included in TCP (n + 1).
  • t (n + 1) is calculated based on PCR (n + 1), I and R (n).
  • FIG. 9 is a flowchart illustrating a method of operating a transmitting side according to an embodiment of the present invention.
  • the transmitting side is a streaming service transmitting apparatus
  • the receiving side is a streaming service receiving apparatus.
  • Operations 910 to 970 show a streaming service transmission method performed by a transmitting side.
  • one or more TS packets are mapped in TCP (n + 1). That is, one or more TS packets are loaded into the payload of TCP (n + 1).
  • operation 920 it is checked whether a TS packet containing PCR information in the TS header exists in the payload of TCP (n + 1). That is, it is checked whether there is a TS packet including PCR information in the TS header among one or more TS packets loaded in the payload of TCP (n + 1).
  • operation 930 If a TS packet including PCR information in the TS header exists in the payload of TCP (n + 1), operation 930 is performed, otherwise operation 960 is performed.
  • TCP time stamps to be described later are not calculated and recorded for TCP (n + 1).
  • PCR (n + 1) is set.
  • Initial PCR information in one or more TS packets in the payload of TCP (n + 1) is set to PCR (n + 1) of TCP (n + 1).
  • a timestamp TCPt (n + 1) of TCP (n + 1) is calculated.
  • the timestamp TCPt (n + 1) is the expected arrival time to the receiver side of the theoretically expected TCP (n + 1).
  • the timestamp of TCP may represent a time reference value for streaming data in the TCP packet stream.
  • the visual reference value may be calculated based on the STC from which the PCR is derived.
  • TCPt (n + 1) is a time corresponding to t (n + 1) of FIG. 8 (that is, the time at which the first byte of the payload of TCP (n + 1) is estimated to arrive at the receiving side). It may be expressed as a clock count value of the STC using.
  • TCPt (n + 1) like PCR (n + 1), can be divided into a base clock TCPt_base (n + 1) and an extension clock TCPt_ext (n + 1). That is, TCPt (n + 1) may include a base clock and an extended clock.
  • the base clock can use smaller frequency units than the extended clock.
  • the base clock may be expressed in units of 90 KHz (ie SCF / 300).
  • the extended clock may be expressed in units of 27 MHz (ie SCF).
  • the timestamp TCPt (n + 1), the base clock TCPt_base (n + 1) and the extended clock TCPt_ext (n + 1) can be calculated according to Equation 4 below.
  • TCPt (n + 1), TCPt_base (n + 1) and TCPt_ext (n + 1) may be calculated based on the system clock frequency and t (n + 1).
  • TCPt (n + 1) has a precision of 27 MHz.
  • TCPt_base (n + 1) may be represented by a total of 33 bits in length
  • TCPt_ext (n + 1) may be represented by a total of 9 bits in length
  • TCPt_base (n + 1) and TCPt_ext TCPt (n + 1), which is the sum of (n + 1), can be represented by a total length of 42 bits as in PCR.
  • TCPt (n + 1) is reset to zero again when TCPt (n + 1) has reached its maximum value can be used for the calculation of TCPt (n + 1).
  • the maximum error that occurs when the time is expressed in a clock of 90 KHz is to be.
  • TCPt_base (n + 1) may be used when the accuracy of the arrival time delay variation may exceed 11 ⁇ s.
  • the maximum error of 11 ⁇ s that can occur due to the use of only TCPt_base (n + 1) is included in the arrival time delay variation itself.
  • the length of the bits of TCPt (n + 1) may be determined based on the precision of the arrival time delay variation required by the receiver.
  • TCPt (n + 1) is written into TCP (n + 1).
  • TCPt (n + 1) may be stored in the header of TCP (n + 1) and may be stored in the options field in the header.
  • TCPt (n + 1) A specific method of storing TCPt (n + 1) in the header of TCP (n + 1) is described in detail below with reference to FIG.
  • TCPt (n + 1) is sent to the receiving side.
  • n is incremented by 1 for the next TCP packet to be processed, and operation 910 is performed again.
  • the transmitting side for generating and transmitting the TCP packet may be a streaming server and may be an HTTP streaming server.
  • the receiving side receiving the TCP packet may be a client or a terminal. It is possible to receive a TCP packet through HTTP streaming on the receiving side.
  • FIG. 10 illustrates a basic structure of a header 310 of a TCP packet and an extension field that may be selectively used according to an embodiment of the present invention.
  • the header 310 of the TCP packet is 20 bytes in size.
  • the TCP packet may add header information up to 32 bytes by utilizing the Options field 1010 to include additional information.
  • Whether the options field 1010 is added or not may be specified through a data offset 1020 (indicating the length of the header).
  • the calculated TCP timestamp value may be added to the options field 1010.
  • the options field can be largely composed of three fields, such as Kind, Length, and Data.
  • the 1-byte type field indicates the purpose.
  • the 1-byte length field indicates the total size of the options field corresponding to the current type field.
  • the data field contains the actual data needed for the purpose indicated by the kind field.
  • FIG. 11 is a structural diagram of a transmitting side 1100 according to an embodiment of the present invention.
  • the transmitting side 110 is a streaming service transmitting apparatus (eg, a streaming server).
  • the transmitting side 1100 includes a TS packet load unit 1110, a PCR setting unit 1120, a time stamp calculation unit 1130, a time stamp recording unit 1140, and a transmission unit 1150.
  • the TS packet load unit 1110 performs operations 910, 920, and 970. For example, the TS packet load unit 1110 loads one or more TS packets in the payload of the TCP packet.
  • the PCR setting unit 1120 performs operation 930. For example, the PCR setting unit 1120 sets first PCR information in one or more TS packets in a payload to PCR of a TCP packet.
  • the time stamp calculator 1130 performs operation 940. For example, the time stamp calculator 1130 calculates a time stamp of the TCP packet based on the PCR.
  • the time stamp recording unit 1140 performs operation 950.
  • the time stamp recording unit 1140 records the calculated time stamp in the TCP packet.
  • the transmitter 1150 performs operation 960. For example, the transmitter 1150 transmits a TCP packet to the terminal.
  • FIG. 12 is a flowchart illustrating a method of operating a receiving side according to an embodiment of the present invention.
  • the operations 1210 to 1295 illustrate a streaming service receiving method performed by the receiving side.
  • the receiving side receives the TCP packet transmitted by the transmitting side described with reference to FIG.
  • the receiving side utilizes the value of TCP timestamp TCPt (n + 1), which is carried in a TCP packet.
  • the receiver receives the TCP packet through the IP network. Estimates of the delay variation of the arrival time experienced.
  • n + 1 th packet receives a TCP packet.
  • the received TCP packet is the n + 1 th packet.
  • the n th packet received before the n + 1 th packet is referred to as a second TCP packet.
  • the clock count of the first TCP packet described below may be referred to as a first clock count.
  • TCPa (n + 1) is a clock count value of the receiving STC at the moment when the n + 1th TCP packet is received.
  • TCPa (n + 1) is the clock count of the first TCP packet.
  • STC may be based on 27Mhz.
  • TCPt (n + 1), which is a TCP timestamp, is present in the received TCP packet.
  • TCPt (n + 1) is the time when the receiving side expected by the transmitting side of the TCP packet receives the TCP packet.
  • the TCP timestamp may be in the options field in the header of the TCP packet. Thus, it can be checked whether a TCP timestamp exists in the options field in the header of the received TCP packet.
  • TCPt (n + 1) is present in the received TCP packet, operation 1240 is performed, otherwise operation 1260 is performed.
  • TCPt (n + 1) is extracted from the TCP packet.
  • TCPt (n + 1) may be extracted from the options field in the header of the TCP packet.
  • network jitter N (n + 1) experienced by TCP (n + 1) is calculated.
  • the network jitter N (n + 1) may be calculated according to Equation 5 below.
  • Network jitter N base (n when TCPt_base (n + 1) of length 33 bits with a 90 KHz precision is recorded in the TCP packet TCP (n + 1) (or optional fields of the header of the TCP packet) as a timestamp. +1) may be calculated according to Equation 6 below.
  • TCPa_base (n + 1) is a base clock of TCPa (n + 1) and has a unit of 90 KHz.
  • the network jitter N base (n + 1) according to Equation 6 has an error in the range as shown in Equation 7 below, compared to the network jitter N (n + 1) according to Equation 5.
  • TCP (n) is a packet that arrives before TCP (n + 1) arrives at the receiving end. In other words, TCP (n) and TCP (n + 1) are neighboring packets.
  • TCPt (n) is the timestamp of TCP (n).
  • TCPa (n) is the clock count value of the receiving STC at the moment when TCP (n) is received.
  • N (n) is the network jitter experienced by TCP (n).
  • an inter packet spacing D (n + 1) between TCP (n) and TCP (n + 1) is calculated.
  • TCP time stamps of 42 bits in length with a precision of 27 MHz 1) may be calculated according to Equation 8 below.
  • TCPt_base (n + 1) and TCPt_base (n), which are 33-bit long TCP timestamps with a precision of 90 KHz, are used
  • the arrival time interval D base between TCP (n) and TCP (n + 1) (n + 1) may be calculated according to Equation 9 below.
  • An arrival time interval between TCP (n) and TCP (n + 1) obtained based on Equations 8 and 9 may be represented by D (n + 1).
  • arrival interval jitter and variance and de-jitter buffering times of the arrival interval jitter are calculated.
  • Arrival interval jitter is the arrival interval delay variation between successive arriving packets.
  • the arrival interval jitter J (n + 1) of TCP (n + 1) may be calculated according to Equation 10 below by a moving average.
  • J (n) is the arrival interval jitter of TCP (n).
  • ⁇ (0 ⁇ ⁇ ⁇ 1) is the first smoothing factor. ⁇ can adjust the reflectance ratio of recently updated network jitter.
  • ⁇ (n + 1) is the variance for J (n + 1).
  • v (n + 1) is the arrival interval variance of J (n + 1).
  • v (n) is the variance over J (n).
  • ⁇ (n + 1) may be calculated by the moving average according to Equation 11 below.
  • ⁇ (0 ⁇ ⁇ ⁇ 1) is the second smoothing factor.
  • ⁇ can control the rate of response to variations between recently updated network jitter and existing average jitter.
  • B (n + 1) is the de-jitter buffering time for effectively absorbing J (n + 1).
  • Equation 12 B (n + 1) may be calculated according to Equation 12 below.
  • K is a parameter for reflecting the fluctuation range of the variance ⁇ (n + 1), which is a deviation from the average arrival interval jitter value in determining the de-jitter buffering time.
  • B (n + 1) is calculated based on N (n + 1) and D (n + 1).
  • B (n + 1) is calculated based on TCPt (n + 1), TCPt (n), TCPa (n + 1) and TCPa (n).
  • K The larger the value of K, the greater the probability that an unusually large fluctuation that can occur in a particular situation in the network can be absorbed by de-jitter buffering.
  • the initial playout delay time for the first packet to arrive at the receiver is experienced through de-jitter buffering. Can be determined.
  • an initial reproduction time of the progressive streaming service may be determined in consideration of absorption of network jitter occurring during transmission of a packet through the IP network.
  • operation 1260 it is checked whether the analysis time for de-jitter buffering time determination has elapsed.
  • TCP packets In order to estimate the appropriate de-jitter buffering time by utilizing the timestamp of the received TCP packet (ie TCP (n + 1)), TCP packets must be received and analyzed for some sufficient time. The time required for such reception and analysis is called the analysis time T.
  • Sufficient time for the transmission of a TCP packet to a stable packet transmission and reception state may be sufficient as analysis time.
  • a time of about 3 to 4 seconds is sufficient as the analysis time.
  • Operation 1280 is performed when analysis time T for determining the de-jitter buffering time has elapsed, otherwise operation 1270 is performed.
  • n is incremented by 1 to process the next packet, and operation 1210 is performed.
  • operation 1280 it is checked whether the calculated de-jitter buffering time B (n + 1) is greater than the analysis time T. Operation 1290 is performed if the calculated de-jitter buffering time B (n + 1) is greater than analysis time T; otherwise operation 1295 is performed.
  • the calculated de-jitter buffering time B (n + 1) is determined as the final de-jitter buffering time t dj .
  • analysis time T is determined as the final de-jitter buffering time t dj .
  • Equation 13 the larger of B (n + 1) and T is determined as the final de-jitter buffering time t dj, as shown in Equation 13 below.
  • Operation 1290 or operation 1295 is performed, followed by operation 1270.
  • FIG. 13 illustrates a T-STD buffer model including de-jitter buffering according to an embodiment of the present invention.
  • the T-STD buffer model provides de-jitter buffering to absorb jitter that occurs when receiving progressive streaming service over an IP network, based on the arrival interval jitter according to the embodiment of the present invention described above.
  • the de-jitter buffering time B (n + 1) or t dj calculated in FIG. 12 may be set as the buffering time of the T-STD of the receiving side.
  • IP network de-jitter buffering to absorb network jitter takes place inside the file buffer at the receiving end.
  • TS packets delivered to the existing T-STD buffer model may be processed according to the operating principle of the existing T-STD buffer model.
  • FIG. 14 is a structural diagram of a receiving side 1400 according to an example of the present invention.
  • the receiving side 1410 is a streaming service receiving apparatus (eg, a terminal).
  • the receiving side 1400 includes a receiving unit 1410, a clock count measuring unit 1420, a timestamp extracting unit 1430, and a de-jitter buffering time calculating unit 1440.
  • the receiver 1410 may include a T-STD buffer 1450.
  • the receiver 1410 performs operation 1210. For example, the receiver 1410 receives TCP (n + 1) and TCP (n).
  • the clock count measurer 1420 performs operation 1220.
  • the clock count measuring unit 1420 measures the clock count TCPa (n + 1) at the moment when TCP (n + 1) is received, and measures the clock count TCPa (n) at the moment when TCP (n) is received. do.
  • the time stamp extractor 1430 performs operations 1230 and 1240. For example, the time stamp extractor 1430 checks whether TCPt (n + 1) exists in TCP (n + 1), and extracts TCPt (n + 1) from TCP (n + 1).
  • the de-jitter buffering time calculator 1440 performs operations 1244 to 1260, 1280, 1290, and 1295.
  • the de-jitter buffering time calculator 1440 may calculate the de-jitter buffering time B (n + 1) based on TCPt (n + 1), TCPt (n), TCPa (n + 1), and TCPa (n). Or t dj .
  • the de-jitter buffering time calculator 1440 may include N (n + 1), N (n), N base (n + 1), J (n + 1), J (n), and D (n + 1). ), ⁇ (n + 1), ⁇ (n) and the like.
  • 15 illustrates a delay of media playback time due to de-jitter buffering configured according to an embodiment of the present invention.
  • the temporal modeling shown in FIG. 15 is as follows 1) to 3).
  • the first TCP packet arriving at the file buffer stays in the file buffer for the de-jitter buffering time determined by the embodiment of the present invention described above.
  • TS packets included in the first packet are transferred to the existing T-STD buffer model.
  • the x axis represents the time at the transmitting side and the receiving side
  • the y axis represents the index of the transmitted or received TCP packet.
  • time 1510 at which the sender transmitted TCP packets, the time 1520 at which the receiver received TCP packets, and the delivery deadline 1530 in the conventional T-STD buffer model are shown. Also shown is a time point 1540 where TS packets in the first TCP packet are delivered to a conventional T-STD buffer for playback.
  • the end-to-end transmission delay is the difference between the time at which the TCP packets were sent 1510 and the time at which the TCP packets were received 1520, and the de-jitter buffering time (ie, the playback delay time) is the TCP packet. Difference between the reception time 1520 and the propagation deadline 1530 in a typical T-STD buffer model.
  • de-jitter buffering time should be considered integrally with the T-STD buffer model of the receiver in order to provide a high quality progressive streaming service.
  • the operation principle of the existing T-STD buffer model can be followed as it is.
  • the T-STD buffer model developed as a standard can be utilized without modification in the method and apparatus according to embodiments of the present invention.
  • Method according to an embodiment of the present invention is implemented in the form of program instructions that can be executed by various computer means may be recorded on a computer readable medium.
  • the computer readable medium may include program instructions, data files, data structures, etc. alone or in combination.
  • Program instructions recorded on the media may be those specially designed and constructed for the purposes of the present invention, or they may be of the kind well-known and available to those having skill in the computer software arts.
  • Examples of computer-readable recording media include magnetic media such as hard disks, floppy disks, and magnetic tape, optical media such as CD-ROMs, DVDs, and magnetic disks, such as floppy disks.
  • Examples of program instructions include not only machine code generated by a compiler, but also high-level language code that can be executed by a computer using an interpreter or the like.
  • the hardware device described above may be configured to operate as one or more software modules to perform the operations of the present invention, and vice versa.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)

Abstract

L'invention concerne un dispositif et un procédé d'émission et de réception de service de diffusion en mode continu. Un sautillement d'intervalle d'arrivée est une une variation de temps de retard d'arrivée entre des paquets TCP qui arrivent dans un dispositif recevant un service de diffusion en mode continu. Le dispositif recevant un service de diffusion en mode continu détermine un temps de mise en mémoire tampon de suppression de sautillement pour absorber un sautillement de réseau avant que les paquets TS dans un paquet TCP soient appliqués à un modèle de mémoire tampon T-STD. Un temps de retard de reproduction initial du dispositif recevant un service de diffusion en mode continu est déterminé par le temps de mise en mémoire tampon de suppression de sautillement.
PCT/KR2011/007827 2010-10-20 2011-10-20 Dispositif et procédé d'émission/réception de service de diffusion en mode continu WO2012053834A2 (fr)

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US13/880,941 US20130282871A1 (en) 2010-10-20 2011-10-20 Streaming service transmitting/receiving device and method

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KR10-2010-0102574 2010-10-20
KR1020110030790A KR101180540B1 (ko) 2010-10-20 2011-04-04 스트리밍 서비스 송/수신 장치 및 방법
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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113949896A (zh) * 2021-09-30 2022-01-18 中央广播电视总台 Ip媒体流处理方法、装置、计算机设备及可读存储介质

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20050071491A1 (en) * 2003-09-27 2005-03-31 Lg Electronics Inc. Multimedia streaming service system and method
JP2005522916A (ja) * 2002-04-09 2005-07-28 コーニンクレッカ　フィリップス　エレクトロニクス　エヌ　ヴィ ダウンロードとストリーミングを組み合わせる伝送方法
KR20060114080A (ko) * 2005-04-27 2006-11-06 삼성전자주식회사 멀티미디어 스트리밍 서비스 시스템 및 방법
US20090271525A1 (en) * 2006-04-24 2009-10-29 Electronics And Telecommunications Research Instit Rtsp-based progressive streaming method

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2005522916A (ja) * 2002-04-09 2005-07-28 コーニンクレッカ　フィリップス　エレクトロニクス　エヌ　ヴィ ダウンロードとストリーミングを組み合わせる伝送方法
US20050071491A1 (en) * 2003-09-27 2005-03-31 Lg Electronics Inc. Multimedia streaming service system and method
KR20060114080A (ko) * 2005-04-27 2006-11-06 삼성전자주식회사 멀티미디어 스트리밍 서비스 시스템 및 방법
US20090271525A1 (en) * 2006-04-24 2009-10-29 Electronics And Telecommunications Research Instit Rtsp-based progressive streaming method

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113949896A (zh) * 2021-09-30 2022-01-18 中央广播电视总台 Ip媒体流处理方法、装置、计算机设备及可读存储介质
CN113949896B (zh) * 2021-09-30 2023-11-17 中央广播电视总台 Ip媒体流处理方法、装置、计算机设备及可读存储介质

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