Abstract: A telecommunication apparatus (100, 200) enabled for high-speed packet access is disclosed. The apparatus (100, 200) is arranged to operate according to a reduced and a further reduced mode of transmission of dedicated physical control channel transmission, and having a data transmission controller (102, 202) arranged to control sporadic data transmissions. The data transmission controller (102, 202) is arranged to determine if omission of a sporadic data transmission will significantly degrade performance, and if not, disable transmission of that data transmission. A method of controlling sporadic data transmissions for such an apparatus is also disclosed, as well as a computer program for implementing the method.

Abstract: A terminal (30, 30B) receives transmissions in a form of a media stream. The terminal comprises a jitter buffer (40) which receives data comprising the media stream and a buffer manager (80). The buffer manager (80) makes a selection between plural playback pointers as an operative playback pointer from which the data comprising the media stream is played out of the jitter buffer. In an example implementation, the buffer manager (80) updates at least one of the plural playback pointers. The manner and timing of the updating of the least one of the plural playback pointers can occur in any of a variety of ways. The terminal (30, 30B) can take various forms, and may be (for example) either a wireless terminal which receives the media stream across a radio interface, or a wireline terminal.

Abstract: In a method and a system a good spread of users per code is obtained by providing a value related to the current load for each code in the system. The estimated load value can be used by the network to assign the HS-PDSCH code for users in HS-SCCH less operation, and may also be used in the dynamic transmission process, i.e. which codes are to be used by which user during this and future TTIs. If one code is over a given utilization threshold, where the threshold is set in order to avoid code blocking of VoIP users, enough users can be re-assigned to a different code. In the case that all of the current codes used in the HS-SCCH less operation model are over the utilization threshold, yet another code can be made available for HS-SCCH less operation.

Abstract: The present invention relates to a method of reducing setup delay for an uplink message from a user terminal (UT) in a delay sensitive service in a radio telecommunications system, such as a push to talk service (PoC), by predicting that delay sensitive data is about to be transmitted, sending, as a response to the prediction, a connection setup signal from the terminal to a basestation subsystem (BSS) in order to set up an early uplink radio connection, and transmitting the delay sensitive data via the early uplink connection. There is also provided a user terminal (UT) and a radio telecommunications system.

Abstract: In a method of enabling improved group communication for a plurality of mobile terminals MT participating in a multicast session in a communication system comprising a plurality of cells and at least two base station systems with corresponding cell coverage, reducing S1 mandatory paging signalling from said mobile terminals and said network, by maintaining said network and said mobile terminals in an active state, and limiting S2 cell update signalling in response to one of said plurality of mobile terminal moving from one cell to a new cell.

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: Method and arrangement in a first node, for generating a first and a second correctness target value. The generated correctness target values are used by a power control of the radio signals sent from a second node to be received by the first node. The radio signals are sent over at least a first channel and a second channel. The method comprises establishing a difference between the obtained first quality value of the first channel with a first quality target value of the first channel and generating a first correctness target value. The method also comprises the step of establishing a second difference between the obtained second quality value of the second channel with a second quality target value of the second channel and generating a second correctness target value.

Abstract: Adaptive redundancy is implemented (either switched on or off) for a voice over internet protocol (VoIP) packet connection by a basic two step approach. A first step of adaptive redundancy implementation involves monitoring relevant sources or indicators to determine when the need for redundancy (e.g., redundancy coding) may arise. A second step of adaptive redundancy implementation involves the triggering of redundancy implementation events when the monitored sources reach or obtain certain threshold value(s) (threshold). The first and second steps of redundancy implementation can be repeated as needed. Scenarios of adaptive redundancy implementation include both network-initiated implementations (e.g., implementations initiated by a radio access network (RAN)) and mobile-initiated implementations (e.g., implementations initiated by a wireless station).

Abstract: The present invention relates to a group communication solution in a wireless communication network comprising at least one base station (5) and a plurality of terminals (1, 2, 3, 4) communicating wirelessly with the base station and where the terminals has at least partly identical identity references in order to be part of a multicast protocol push to talk group.

Abstract: The invention involves uplink access management for user terminals (210, 220, 230, 240, 250, 260, 270) present in a same first cell (15) of a communication system (1) and participating in a communication session involving communication of user data on a common downlink channel to the terminals (210, 220, 230, 240, 250, 260, 270). When the terminals (210, 220, 230, 240, 250, 260, 270) want to respond to the user data transmission, they first have to request uplink access. According to the invention, at least one of the terminals (220, 230, 240) requests this uplink access using a request channel associated with a second (25, 35, 45) cell different from the current first cell (15). This reduces the risk of collision of uplink access requests on the request channel by distributing the requests over several cells (15, 25, 35, 45) instead of a single cell (15).

Abstract: Within a cell area, an RBS typically provides communication services to UEs. During operation, the UE adjusts its transmission power according to a transmission parameter value such as the E-TFCI. The RBS provides a predetermined number of reference points for E-TFCI that covers the limited range. For any intermediate values not specifically provided, the UE calculates the SIR through interpolation. The RBS predicts a range of transmission power likely to be used by the UE and provides the reference points based on the prediction. The prediction is performed based on operational measurement.

Abstract: The present invention is directed towards providing event-triggered communication in a half duplex communication environment of a wireless communication network (10). A first end terminal (12) sends a selection of an end terminal (14) in a group of terminals to act as an event notifying device to a server (22). The event notifying device (14) compares an event with a threshold and sends an event notifier to the server if the event exceeds the threshold. If it receives the event notifier, the server prioritises data from the event notifying device over ongoing communication sessions for other devices in the group, receives data from the event notifying device associated with the event notifier and transports data indicative of the event to other devices of the group, in order to inform them of the event.

Abstract: The present invention is directed towards compressing and decompressing data transmitted between an end terminal (12) and a network server (20) controlling compression in communication sessions involving the end terminal. States related to the end terminal that are created and used for compressing and decompressing data transmitted between the end terminal and the network server when the end terminal is first registered to the first network server are saved as the end terminal deregisters from the first network server. When the end terminal again registers to the network server these saved states are directly used. In this way it is possible to start compressing data using the saved states immediately after a following registering for providing a more effective dynamic compression.

Abstract: A Multimedia system and a method in a Multimedia system where a first user with user equipment (1) uses a Push to talk service (PTT) to communicate with at least one second subscriber's user equipment (2) using said Push to talk service (PTT), wherein a PTT-server (3) providing the PTT-service interacts with said first and second subscriber's user equipment (1, 2) and exchanges information concerning relevant parameter values for each user equipment (1, 2), and each user equipment (1, 2) adapting its application settings during the operation of said PTT-service, whereby degradation of the perceived quality is prevented

Abstract: A telecommunications network comprises a sending entity (22) and a receiving entity (24) which communicate over a radio interface (26). The sending entity (22) comprises a compressor (40) and a message formatter (34). The compressor (40) in turn comprises a sending entity compression dictionary (60) and dictionary update decision logic (62). The compressor (40) compresses a message to obtain a compressed message to be sent over the radio interface (26) to the receiving entity (24) and also determines a compression factor for the message. The compression factor is based on similarity of the message to contents of the sending entity compression dictionary. The dictionary update decision logic (62) makes a decision whether the compression factor justifies updating of a receiving entity decompression dictionary (70). The message formatter (34) includes, in the compressed message, an update indication regarding the decision.

Abstract: A method of operating a base station node (21) comprises obtaining an indication of degradation of a radio link connection between the base station node and the user equipment unit (UE) which is carried by the a high speed packet channel (HSDPA). In accordance with the distance indication, the transmission priority for the radio link connection is modified. Preferably the indication of degradation is a distance indication of distance of a user equipment unit (23) involved in the radio link connection from the base station node (21). The transmission priority can be modified for one or both of downlink and uplink transmissions for the user equipment unit (UE). Preferably the transmission priority is modified for the radio link connection, e.g., for the user equipment unit (UE), as a function of the indication.

Abstract: The present invention relates to a method and a user equipment arranged to communicate with at least a second user equipment in a VoIP service data transmission in a wireless communication system using a VoIP service, provided by an application server. The method comprises the steps of: receiving transmissions in form of a media stream from the at least one second user equipment; storing the data of the media stream; detecting whether an interruption of said transmissions having at least a minimum length has occurred during the VoIP service data transmission or expecting that an interruption of said transmissions having at least a minimum length will occur during the VoIP service data transmission, and if such an interruption is detected or expected, using a non-normative playout rate of the data at playout, thereby obtaining a more efficient interactivity in the user communication.

Abstract: Adaptive redundancy is implemented (either switched on or off) for a voice over internet protocol (VoIP) packet connection by a basic two step approach. A first step of adaptive redundancy implementation involves monitoring relevant sources or indicators to determine when the need for redundancy (e.g., redundancy coding) may arise. The monitored sources can be one or more of a transmission buffer(s); a receiving buffer(s); sender and/or receiver statistics. An example of such statistics are the statistics which can be obtained from RTCP protocol, e.g., number of packets/data bytes sent, number of packets/data bytes lost, jitter, etc. A second step of adaptive redundancy implementation involves the triggering of redundancy implementation events when the monitored sources reach or obtain certain threshold value(s) (threshold).

Abstract: The present invention relates to management of a state memory in a communications unit. The state memory then stores states that are used in message-based communication with external units in a communications system. The data message communicated between the units are defined into multiple message classes. Furthermore the state memory is divided into multiple memory portions, where each such memory portion is assigned to store states associated with a specific message class. This will prevent overwriting important states with less useful states.