摘要:
A method of signaling and negotiation between a client and a server in a multimedia streaming service regarding the adaptation of the data delivery process. In order to make sure that the client supports the adaptation mechanisms or capabilities to be used in data delivery, one of the parties provides information indicating the adaptation mechanism or capability that it supports to the other party. Upon receiving the information, the other party uses well-defined RTSP response to indicate the support of that mechanism or capability. Either the server or the client can initiate the negotiation. The implementation of the signaling and negotiation covers an AVPF usage, RTP Retransmission Playload Format usage, and an SPTP usage in a particular 3G PSS session.
摘要:
A method to provide to a sender (20) of RTP packets the actual receiver (50) buffer (54) fullness level in a receiver of the RTP packets at a certain time instant represented as remaining playout duration in time. The receiver sends in an RTCP report (26) the sequence number of a selected RTP packet in the receiver buffer and the time difference between the scheduled playout time of this packet and the current time. Based on this timing information, the sender calculates the amount of time it would take for the receiver buffer to empty if the receiver continues to playout at normal speed and no further RTP packets arrive to the receiver buffer. This receiver buffer fullness level information can be used at the sender to adjust the transmission rate and/or nominal playout rate of the RTP packets in order to maintain a targeted receiver buffer fullness level.
摘要:
A method to provide to a sender (20) of RTP packets the actual receiver (50) buffer (54) fullness level in a receiver of the RTP packets at a certain time instant represented as remaining playout duration in time. The receiver sends in an RTCP report (26) the sequence number of a selected RTP packet in the receiver buffer and the time difference between the scheduled playout time of this packet and the current time. Based on this timing information, the sender calculates the amount of time it would take for the receiver buffer to empty if the receiver continues to playout at normal speed and no further RTP packets arrive to the receiver buffer. This receiver buffer fullness level information can be used at the sender to adjust the transmission rate and/or nominal playout rate of the RTP packets in order to maintain a targeted receiver buffer fullness level.
摘要:
The invention relates to a method for transmitting video images between video terminals (1, 1') in a data transmission system. Video images comprise frames (T0, T1, ..., T9), which are divided into slices (S1-S8, SX). Every frame (T0, T1, ..., T9) comprises at least two slices (S5, S3, S6; S1, SX, S2; S7, S4, S8) which are at least partly adjacent to each other, and consecutive frames (T0, T1, ..., T9) have corresponding slices (S5, S1, S7; S3, SX, S4; S6, S2, S8). The slices (S1-S8, SX) are interleaved into packets, and the packets are transmitted. The interleaving is performed in such a way that adjacent slices (SX, S1; SX, S2) in the same frame (T1) are transmitted in different packets, and that corresponding slices (S5, S1, S7; S3, SX, S4; S6, S2, S8) of two consecutive frames (T0, T1, T2) of video images are transmitted in different packets. Then every packet comprises only such slices which are other than adjacent to each other in the same frame and other than corresponding slices of two consecutive frames.
摘要:
The present invention relates to a method and a communication system for transmission of multimedia streams. Multimedia streams are transmitted in the communication system from a sending communication device to a receiving communication device at least partly via a wireless communication network. Information about the multimedia stream is transmitted to the receiving communication device comprising at least one parameter of the transmission of the multimedia stream for reservation of network resources. The parameter is the maximum bit rate which is needed for the transmission or the maximum service data unit size to be used in the transmission. It is also possible that both the mentioned parameters will be transmitted as attributes of the session description protocol. In an advantageous embodiment the receiving communication device informs the sending communication device about the QoS profile parameters which the wireless communication network granted for the transmission.
摘要:
A method and system for adaptively controlling level of a receiver buffer in a client in a multimedia streaming network. The multimedia streaming network has a server for providing streaming data to the client. The server is responsible for adapting the transmission rate to the reception rate or congestion control, and for adapting the sampling rate to the transmission rate. Thus, the server manages the shift and keeps it within the rate adaptation operating range. The client is responsible for compensating for the packet transfer delay variation, which is also known as network jitter. The client is also responsible for setting parameters of the server rate adaptation operating range. The client chooses and sends the shift parameters to the server, but it is up to the server to adapt its encoding rate or transmission rate when responding to the parameters.
摘要:
A method and device for enabling packet transfer delay compensation in multimedia streaming. In order to enable a streaming server to optimally operate its rate-control and rate-shaping algorithms to compensate for packet transfer delay variation, information indicative of jitter buffering capabilities of the streaming client is conveyed to the streaming server. The information contains the client's chosen pre-decoding parameters so that the client's jitter buffering capabilities can be determined by the server based on the difference between the client's chosen pre-decoding parameters and the pre-decoding buffering parameters provided by the streaming server.
摘要:
The present invention relates to a method for transmitting video information (fig. 5), in which a bitstream (510,520) is formed comprising a set of frames (512,513,515,516,518 and 522,523,525 and 528) comprising macroblocks. At least one switching frame (524) is formed into the bitstream and macroblocks of the switching frame are arranged into a first and a second group of macroblocks, each macroblock of the first group are encoded by a first encoding method (fig. 5, intra) to provide a switching point for continuing the transmission of video information with another bitstream formed from the video information.
摘要:
A method for signaling and negotiation between a resource-limited client and a server in a multimedia streaming service regarding packet data delivery. In order to avoid dropping packets at the client side due to its maximum packet rate capability, the client signals to the server declaring the maximum packet rate capability. This capability can be signaled to the client via a capability exchange mechanism or using a multimedia streaming protocol. The client inserts a parameter indicative of the maximum packet data rate capability in a request sent to server. It is up to the server to take the necessary action and make the packet delivery rate adjustment.
摘要:
A method used for communicating packets or other information units from a server (11) to a client (15) over a communication path including a packet scheduler (14-1) followed by a bottleneck path (14-2), with the packet scheduler (14-1) having a buffer for holding packets so as not to transmit over the bottleneck path (14-2) at a rate either too high or too low, the method providing that the server (11) communicate to the client (15) the last packet sent to the client (11) and do so via a mechanism by which the sender information is communicated over the bottleneck path (14-2) ahead of substantially all packets already in the buffer. The client (11) then uses the sender information to provide, as needed, information useful in adapting the rate at which the packets arrive at the packet scheduler (14-1).