Abstract: Methods of transmitting information relating to a quality of service are shown, which information is to be transmitted in at least one direction between a first device 30 and a second device 20. A first method comprises at least at one of the devices 20, 30 assembling a protocol message containing information other than the information relating to a quality of service and attaching the quality of service related information to the protocol message. A second method comprises forming the information relating to a quality of service inside at least one of a header field and an attribute of a protocol message. The disclosure relates equally to corresponding software codes, devices 20, 30, network elements and systems.

Abstract: The invention discloses a method of improving the playback of streamed media on a client device by overcoming problems caused by variations in the transmission delay of packets due to network and transport protocol operation and variations in encoding/server specific delays. In an embodiment of the invention, a client device has a decoder (120) and a predecoder buffer (110) which receives streamed packets from source server via a packet network. The pre-decoder buffer is variable in size and has a variable initial buffering time for receiving the transmitted packets from the source server prior to decoding in the decoder. The initial buffering time and pre-decoder buffer size can be dynamically adapted for improved playback performance by the source server. In a further aspect of the invention, a post-decoder buffer operates in conjunction with the pre-encoder buffer to reduce decoding-related delay variations.

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: A system and method provide for updating a parameter set in streaming applications. The method includes receiving a first parameter set from a server device at a client device, receiving an updated parameter set having synchronization information at the client device, wherein the updated parameter set is sent from the server device using a Real Time Streaming Protocol (RTSP) method or Session Description Protocol, processing a bitstream received from the server device at the client device using the first parameter set, and processing the bitstream at the client device using the updated parameter set when the synchronization information indicates that the updated parameter set should be used. The synchronization information may include, but is not limited to, a real-time transport protocol timestamp, a normal play time timestamp, network abstraction layer unit time, and a decoding order number.

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.

Abstract: The invention discloses a method of improving the playback of streamed media on a client device by overcoming problems caused by variations in the transmission delay of packets due to network and transport protocol operation and variations in encoding/server specific delays. In an embodiment of the invention, a client device has a decoder (120) and a predecoder buffer (110) which receives streamed packets from source server via a packet network. The pre-decoder buffer is variable in size and has a variable initial buffering time for receiving the transmitted packets from the source server prior to decoding in the decoder. The initial buffering time and pre-decoder buffer size can be dynamically adapted for improved playback performance by the source server.