Abstract: The specification and drawings present a new method, system, apparatus and software product for signaling parameters of multiple buffers by a terminal to a server and determining multiple buffer status of the terminal by the server using these parameters, e.g., for an adequate rate adaptation of multimedia streaming services provided to the terminal by the server. The terminal can be (but is not limited to) a computer, a communication device, a wireless communication device, a portable electronic device, a mobile electronic device, a mobile phone, etc.

Abstract: A method in a communication system, a communication system and a communication device wherein in the method media streams are transmitted from a sending communication device to a receiving communication device at least partly via a wireless communication network. At least one media stream is selected to be transmitted to the receiving communication device. QoS requirements for transmitting the selected at least one media stream are defined, transmission resources are reserved from the wireless communication network for the transmission of at least one media stream, and a setup procedure is performed between the receiving communication device and the sending communication device for activating the one packet data transmission connection.

Abstract: The present invention relates to a method for transmitting video information, in which a bitstream is formed comprising a set of frames comprising macroblocks. At least one switching frame is formed into the bitstream, macroblocks of the switching frame are arranged into a first and a second group of macroblocks, each macroblock of the first group of macroblocks are encoded by a first encoding method to provide a switching point for continuing the transmission of video information with another bitstream formed from the video information; and macroblocks of the second group of macroblocks are encoded by another encoding method. Errors in transmission of video information are reduced by forming at least one SP-encoded frame by predictively encoding the macroblocks; replacing part of the SP-encoded macroblocks with intra encoded blocks; and transmitting the encoded frame containing both predictively and intra encoded macroblocks instead of the SP-encoded frame.

Abstract: A method to provide to a sender of RTP packets the actual receiver buffer 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 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.

Abstract: A method for the client in a multimedia streaming network to control the receiver buffer fullness by way of effectively shifting the transmission clock in the server who provides streaming data to the client. In order to avoid buffer underflow or overflow, the client sends a message to the server requesting the shift of the transmission clock either forward or backward depending on whether the client wants to increase or decrease the receiver buffer. Based on the message, the server adjusts the data transmission amount by effectively shifting its transmission clock. The request from the client can be sent through RTCP or RTSP.

Abstract: The invention relates to a method for packet switched streaming of media, such as video and/or audio, from a streaming server (111) to a mobile client device (101) over an air-interface. The method comprises noticing a change in a downlink air-interface bandwidth, sending from the mobile client device (101) to the streaming server (111) a request for adapting streaming server transmission bit rate, receiving the request at the streaming server (111) and adapting the streaming server transmission bit rate in accordance with the request. According to the method, the request indicates to the streaming server (111) a current downlink air-interface bandwidth and that said adapting of the streaming server transmission bit rate is performed in accordance with said current downlink air-interface bandwidth.

Abstract: Methods, devices, systems, program products and computer-implemented methods for communication of non-interleaved and interleaved packets of streamed data are disclosed. A method of streaming multimedia includes transmitting a signal from a client device to a multimedia server. The signal is indicative of desired settings for the client device to receive packets of streamed data. The desired settings include at least one of an amount of a total buffer memory allocated to buffering of interleaved packets and an indication of acceptability of all packetization types available to the server.

Abstract: Methods, devices, systems, program products and computer-implemented methods for communication of non-interleaved and interleaved packets of streamed data are disclosed. A method of streaming multimedia includes transmitting a signal from a client device to a multimedia server. The signal is indicative of desired settings for the client device to receive packets of streamed data. The desired settings include at least one of an amount of a total buffer memory allocated to buffering of interleaved packets and an indication of acceptability of all packetization types available to the server.

Abstract: The present invention relates to a method for transmitting video information, in which a bitstream is formed comprising a set of frames comprising macroblocks. At least one switching frame is formed into the bitstream, macroblocks of the switching frame are arranged into a first and a second group of macroblocks, each macroblock of the first group of macroblocks are encoded by a first encoding method to provide a switching point for continuing the transmission of video information with another bitstream formed from the video information; and macroblocks of the second group of macroblocks are encoded by another encoding method. Errors in transmission of video information are reduced by forming at least one SP-encoded frame by predictively encoding the macroblocks; replacing part of the SP-encoded macroblocks with intra encoded blocks; and transmitting the encoded frame containing both predictively and intra encoded macroblocks instead of the SP-encoded frame.

Abstract: 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.

Abstract: 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.

Abstract: A method for the client in a multimedia streaming network to control the receiver buffer fullness by way of effectively shifting the transmission clock in the server who provides streaming data to the client. In order to avoid buffer underflow or overflow, the client sends a message to the server requesting the shift of the transmission clock either forward or backward depending on whether the client wants to increase or decrease the receiver buffer. Based on the message, the server adjusts the data transmission amount by effectively shifting its transmission clock. The request from the client can be sent through RTCP or RTSP.

Abstract: A method to provide to a sender of RTP packets the actual receiver buffer 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 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.

Abstract: 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.

Abstract: The invention relates to a method for packet switched streaming of media, such as video and/or audio, from a streaming server (111) to a mobile client device (101) over an air-interface. The method comprises noticing a change in a downlink air-interface bandwidth, sending from the mobile client device (101) to the streaming server (111) a request for adapting streaming server transmission bit rate, receiving the request at the streaming server (111) and adapting the streaming server transmission bit rate in accordance with the request. According to the method, the request indicates to the streaming server (111) a current downlink air-interface bandwidth and that said adapting of the streaming server transmission bit rate is performed in accordance with said current downlink air-interface bandwidth.