摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
Provided are a 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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
An HTTP based video streaming apparatus and method of a mobile communication system is disclosed. A memory, such as a storing disk, stores content files received from a server of a transmitting party, a random access searching unit searches the random access point in the memory and transmits a content file request message to the transmitting server if the random access point does not exist in the memory, and a display unit displays the files from the random access point. Therefore, HTTP streaming service from a random point required by a user can be provided when the streaming has begun, and the random access function can be supported by even if the part required by the user has not yet been transmitted to receiving party.