Abstract: A method and apparatus for an adaptive Hypertext Transfer Protocol (HTTP) streaming service using metadata of media content are provided. The media content may include a sequence of one or more periods. The metadata may include a start time attribute and/or a duration attribute of each of the periods. The metadata may determine a start time of each of the periods, and a start time of each of segments included in each of the periods. A terminal may access segments included in each of the periods, may decode and render data of media content included in the segments, and may play back the media content.

Abstract: A method and an apparatus for maintaining information security in a video multicasting service are provided. The method includes: generating a network abstraction layer unit using received video information; encrypting the network abstraction layer unit of the video information; realtime transport protocol (RTP) packetizing the encrypted network abstraction layer unit of the video information; recording unit format information and field information, included in the network abstraction layer of the video information being stored in a memory, in a header extension field of the RTP header; and transmitting the RTP packet including the encrypted video information to a routing device.

Abstract: A method and apparatus for an adaptive Hypertext Transfer Protocol (HTTP) streaming service using metadata of content are provided. The metadata may include one or more BaseURL elements. Uniform Resource Locators (URLs) of segments forming media may be generated based on the BaseURL elements. Additionally, a URL of a segment may be generated by mapping a BaseURL element among the BaseURL elements to a sourceURL attribute of the segment. Segments indicated by generated URLs may be identical to each other.

Abstract: A method and apparatus for an adaptive Hypertext Transfer Protocol (HTTP) streaming service using metadata of content are provided. The metadata of the content may be efficiently divided for a purpose of use of a terminal based on general media information or specific media information, and may be transmitted to the terminal. A group may include one or more representations of content. The metadata may include a group element, and the group element may provide a summary of attributes of one or more representations included in the group.

Abstract: Provided is an apparatus and method for transmitting and receiving a streaming service. The apparatus and method for transmitting and receiving a streaming service may use a light multimedia file suitable for a Hypertext Transfer Protocol (HTTP) streaming service. Using an HTTP hint track, an International Organization for Standardization (ISO) file generated in a non-interleaved storing scheme that is unsuitable for an HTTP streaming service may be converted into an ISO file based on an interleaved storing scheme that is suitable for an HTTP streaming service. An ISO file may include multiple media tracks for supporting multiple bitrates and resolutions. Chunk data of media tracks that provide multiple multimedia qualities may be effectively acquired, and a light ISO file suitable for an HTTP streaming service may be generated by storing the acquired chunk data in an interleaved scheme.

Abstract: A method and apparatus for an adaptive Hypertext Transfer Protocol (HTTP) streaming service using metadata of content are provided. The metadata may include a minBufferTime attribute indicating a minimum amount of initially buffered media content. A terminal may receive content from a server before playback of the content, and may buffer the content by at least the minimum amount. The metadata may include a range attribute that designates a range of a target indicated by a Uniform Resource Locator (URL). The terminal may receive bytes designated by the range attribute from the URL, and may play back the content using the received bytes.

Abstract: Provided is a real-time streaming protocol (RTSP) based progressive streaming method. The progressive streaming method based on a real-time streaming protocol, wherein a server transfers SDP information and performs streaming of a contents file to a client connected, the method including the steps of: a) if a “pause” request message is received from the client, transmitting a “pause” response message including the number of TCP packets already transferred to the client, and pausing streaming of the contents file; and b) if a “replay” request message including the number of the TCP packet to be transferred is received from the client, transmitting a “replay” response message and restarting transmission of the TCP packets from the next TCP packet to the TCP packet transferred to the client before the contents file is paused.

Abstract: Provided is a method of updating content information associated with a content being serviced in a streaming server group. The content information updating method may include: monitoring whether a new streaming server joins a streaming server group servicing the same category of a content; updating content information associated with the content being serviced in the streaming server group, and transferring the updated content information to a client when the new streaming server joins; and providing a content service via the new streaming server according to a request of the client.

Abstract: A method and an apparatus for maintaining information security in a video multicasting service are provided. The method includes: generating a network abstraction layer unit using received video information; encrypting the network abstraction layer unit of the video information; realtime transport protocol (RTP) packetizing the encrypted network abstraction layer unit of the video information; recording unit format information and field information, included in the network abstraction layer of the video information being stored in a memory, in a header extension field of the RTP header; and transmitting the RTP packet including the encrypted video information to a routing device.

Abstract: A method of supporting synchronization of Scalable Video Coding (SVC) information and Advanced Audio Coding (AAC) information using a Normal Play Time (NPT), the method including: receiving video information using a decoding apparatus; receiving audio information using the decoding apparatus; calculating the NPT of the video information using a Real-time Transport Protocol (RTP) time stamp included in the received video information; calculating the NPT of the audio information using the RTP time stamp included in the received audio information; comparing the NPT of the video information and the NPT of the audio information to calculate a difference value; determining whether the calculated difference value is included in a specific synchronization region; and outputting the audio information and the video information when the calculated difference value is determined to be included in the specific synchronization region.

Abstract: Provided is a streaming service system and method for a universal video access based on scalable video coding (SVC). A gateway to provide a streaming service may include: an SVC content file parser to receive an SVC content file from a streaming server, and to separately extract an SVC network abstraction layer (NAL unit) of a base layer and an SVC NAL unit of a scalable enhancement layer from SVC NAL units that are included in the SVC content file; an SVC NAL selection unit to select an SVC NAL unit corresponding to a client type of each of terminals; and a packet generation unit to transmit the selected SVC NAL unit to a corresponding terminal.

Abstract: Provided are a method for determining the packet type for a Scalable Video Coded (SVC) video bitstream, and a Real-time Transport Protocol (RTP) packetizing apparatus and method using the same. The method for determining a packet type for a Scalable Video Coded (SVC) video bitstream, which includes the steps of: a) deriving temporal and spatial hierarchy information between Network Abstraction Layer (NAL) units from field information defined in the NAL unit headers of scalable layers; b) detecting the type of encoding information by applying combined scalability encoding to the hierarchical structure of the Scalable Video Coding (SVC); and c) determining a Real-time Transport Protocol (RTP) packet type for the corresponding SVC video bitstream by using the derived temporal and spatial hierarchy information between the NAL units and the detected type of encoding information.

Abstract: Provided are a<b> </b>time-stamping apparatus and method for RTP packetization of a SVC coded video, and a RTP packetization system using the same. The time stamping apparatus includes: a NAL unit classifier for checking a header of an input NAL unit and classifying the input NAL units based on a picture property; a first timestamp calculator for calculating a RTP timestamp value for a NAL unit classified as a key picture by the NAL unit classifier; a second timestamp calculator for calculating a RTP timestamp value for a NAL unit classified as a non-key picture by the NAL unit classifier; and a controller for setting a RTP timestamp value for an instantaneous decoding refresh (IDR) picture and controlling the first and second timestamp calculators for calculating a RTP timestamp value of a corresponding NAL unit.

Abstract: Provided is a real-time streaming protocol (RTSP) based progressive streaming method. The progressive streaming method based on a real-time streaming protocol, wherein a server transfers SDP information and performs streaming of a contents file to a client connected, the method including the steps of: a) if a “pause” request message is received from the client, transmitting a “pause” response message including the number of TCP packets already transferred to the client, and pausing streaming of the contents file; and b) if a “replay” request message including the number of the TCP packet to be transferred is received from the client, transmitting a “replay” response message and restarting transmission of the TCP packets from the next TCP packet to the TCP packet transferred to the client before the contents file is paused.