Abstract: In a method of providing a backward and forward compatible speech codec payload format, the following steps are included: providing S10 an RTP package; including S20 payload according to a first codec into the provided RTP package, and appending S50 payload according to a second codec into the provided RTP package. In addition, at least one unused bit is located S30 in the included first codec payload, and the located at least one unused bit is designated S40 as a codec compatibility bit. Finally, the designated at least one codec compatibility bit is utilized S60 to provide an indication of the presence of the appended second codec payload.

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: 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 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: 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: 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.