Abstract: It is a basic idea to determine the characteristics of a jitter profile measured over a plurality of packets, and classify the jitter profile based on the determined characteristics as well as information representative of the particular access used for communication between the sender and the receiver. The classified jitter profile is then matched to an appropriate action for media layer adaptation so that a proper action for media layer adaptation can be initiated.

Abstract: In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

Abstract: User equipment in a wireless communication system monitors scheduling information and locally detects a change in link data rate of the uplink channel based on the monitored scheduling information. In this way, a change in link data rate can be detected directly without significant delay. This direct or early detection of a rate change is then combined with an appropriate system reaction. The information of the detected change in link data rate is preferably utilized for adapting the application data rate of an IP application running in the user equipment. As an alternative, or as a complement, data packets are classified based on relative importance and selected for transfer of information over the uplink channel based on the classification of data packets and in dependence on the detected change in link data rate.

Abstract: In a method of improved session negotiation between first and second clients in a cellular telecommunication system, the two clients negotiating S10 and agreeing upon a first codec type for a session. The first and second clients then initiating the session and exchanging S20 media data frames according to the first codec type. Subsequently, during the initiated session, at least one of the first and second clients providing S30 an indication for a second codec type in at least one subsequent media data frame. Finally, upon receiving and recognizing said indication, the other of said first and second clients switching S40 to the indicated second codec type in a next media data frame, thereby enabling the first and second clients to exchange subsequent media frames utilizing the second codec type.

Abstract: A method of controlling a radio access network entity of a data unit communication network is described, said radio access network entity being involved in providing a transport service to a radio terminal of said data unit communication network, said transport service being characterized by a set of values associated with respective service quality parameters, said method comprising: monitoring (S30) a communication quality at the radio access level of a communication between said terminal and an access point of said radio access network, and setting (S31) a congestion indication marking in data units of said transport service if said communication quality fulfils a radio communication degradation condition.

Abstract: In a method and mobile station for transmitting speech data over a packet data connection a number of speech data frames are bundled into to a bundled frame, which is transmitted over a packet data connection. Hereby the time when the transmitter needs to transmit can be reduced which in turn will reduce the energy required for transmitting speech data. Hereby the battery life time can be increased. The bundling in accordance with the present invention can advantageously be used when transmitting speech data over a High Speed Packet Access. HSPA. connection.

Abstract: In a method and mobile station for transmitting speech data over a packet data connection and video-data over a packet switched connection information about the rendering and capturing clocks for both a Circuit switched (CS) speech connection and a Packet Switched (PS) video connection are determined by a transmitter. The information is transmitted to a receiver and the receiver uses the information to enable synchronization between the speech connection and the video connection.

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.

Abstract: A method of controlling a radio access network entity of a data unit communication network is described, said radio access network entity being involved in providing a transport service to a radio terminal of said data unit communication network, said transport service being characterized by a set of values associated with respective service quality parameters, said method comprising: monitoring (S30) a communication quality at the radio access level of a communication between said terminal and an access point of said radio access network, and setting (S31) a congestion indication marking in data units of said transport service if said communication quality fulfils a radio communication degradation condition.

Abstract: It is a basic idea to determine the characteristics of a jitter profile measured over a plurality of packets, and classify the jitter profile based on the determined characteristics as well as information representative of the particular access used for communication between the sender and the receiver. The classified jitter profile is then matched to an appropriate action for media layer adaptation so that a proper action for media layer adaptation can be initiated.

Abstract: The invention relates to a method for controlling a bit-rate of a session in a packet-switched communications system where multiple sessions are set-up via a shared resource. A current bit-rate of the session is compared to a bit-rate range valid for the session to determine a distance from the position of the current bit-rate to one of the end-points of the bit-rate range. The current bit-rate is then adapted differently depending on the distance to the selected end-point. By controlling the bit-rate of each session that is set-up via the shared resource for applications that provide delay-sensitive services to users in this way the method and corresponding system and sender-receiver arrangements have the effect that rate adaptations can be distributed among the applications so that users share responsibility for rate adaptations.

Abstract: A receiver includes a detector for detecting a change in source of incoming media during an on-going communication session, and means to provide a reset signal in order to reset decoder states of a decoder in response to such a detected change before decoding new incoming media. In this way, a state mismatch can be avoided without the need for several active decoder instances in the receiver, leading to substantial savings with respect to overall complexity, memory usage and power consumption. This also means that media distortions can be eliminated or at least reduced when the decoded media is finally rendered by a player.

Abstract: A method of controlling an adaptable jitter buffer in accordance with the present invention detects a context description of data handled by the adaptable jitter buffer. Thereafter it is determined whether the detected context description is equal to a predetermined context description. If not, jitter buffer adaptation proceeds as normal. Otherwise it is determined whether the current target buffer depth is too low for the detected context description. If not, the target buffer depth is frozen to the current value. Otherwise it is increased and frozen at a higher value that is compatible with the detected context description.

Abstract: In order to efficiently handle the switch between user media and announcement media, a basic step is to first determine a configuration of the user media. Next, a configuration of the announcement media to be presented is determined based on the determined user media configuration. Subsequently, the announcement media is configured according to the announcement media configuration, and the configured announcement media is finally sent to the intended user. In this way, the overall appearance or sound of the announcement will be virtually the same as or at least similar to the overall appearance or sound of the user media, preferably without distortions. This allows the user to perceive the announcement as clearly as possible.

Abstract: In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

Abstract: User equipment in a wireless communication system monitors scheduling information and locally detects a change in link data rate of the uplink channel based on the monitored scheduling information. In this way, a change in link data rate can be detected directly without significant delay. This direct or early detection of a rate change is then combined with an appropriate system reaction. The information of the detected change in link data rate is preferably utilized for adapting the application data rate of an IP application running in the user equipment. As an alternative, or as a complement, data packets are classified based on relative importance and selected for transfer of information over the uplink channel based on the classification of data packets and in dependence on the detected change in link data rate.

Abstract: Packets are selectively marked or dropped when congestion of the radio resources is experienced, the selective marking/dropping being related to or dependent on the probability that a packet will be marked with the relative efficiency of usage of the radio link by the receiver, e.g., dependent upon radio resource usage costs and fairness. For example, packets are marked or dropped based on a user's associated share of the total (or a subset of the) shared radio resources. This share may be expressed in terms of the costs of the resources in terms the user's level of utilization of the shared resources, or in terms of it's fairness with respect to other users sharing the same resources. Thus, the present technology takes into account the distribution of resources usage between receivers contributing to the congested state of the radio network.

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.