An apparatus includes a user input unit, a display unit, a control unit, and a buffer unit. The display unit includes a speed setting menu. The control unit selects a mode from the speed setting menu in response to the selection signal of the user, and controls a compression ratio of a voice codec and a transfer rate of a modem corresponding to a transmission-side radio, and a reception rate of a modem and a restoration rate of a voice codec corresponding to a reception-side radio, based on the selected mode. The buffer unit performs a storage function if there is a difference between the compression ratio of the voice codec and the transfer rate of the modem or if there is a difference between the reception rate of the modem and the restoration rate of the voice codec.

A system for providing sound to at least one user has at least one audio signal source, a transmission unit with a digital transmitter audio data packets from the audio signal source via a wireless digital audio link; at least one receiver unit having at least one digital receiver; and a hearing stimulator responsive to audio signals from the receiver unit. The transmission unit encodes the audio signal as audio data blocks distributed onto at least two audio data packets, one of which is a low-quality packet, and one of which is a high quality packet only a low-quality version of the audio signal being retrievable from the low-quality packets, and a high-quality version of the audio signal being retrievable from both the low-quality packets and the high-quality packets. The low-quality packets and the high-quality packets are transmitted in respective dedicated slots of a multiple access protocol frame.

A base station (BS) for sharing downlink (DL) demodulation reference signals (DMRSs) between the DL data and the DL control signals comprises a storage device for storing instructions and a processing circuit coupled to the storage device. The processing circuit is configured to execute the instructions stored in the storage device. The instructions comprise transmitting a DL control signal on a first layer in a first time-frequency resource to the communication device; transmitting a DL data, associated with the DL control signal on a second layer in the first time-frequency resource and on the first layer and the second layer in a second time-frequency resource, to the communication device; and transmitting a set of DMRSs for the DL control signal and the DL data to the communication device.

An anchoring network device of a telecommunications network can receive a first initiation request of a communication session and determine a second request indicating a different media capability than the first request. The different capabilities can share a payload format. A network gateway can modify headers of packets to change between the capabilities. In some examples, the anchoring device can determine that a codec list of the initiation request corresponds to a rewrite rule. The device can apply the rewrite rule to the request to provide a second request listing a second, different codec. The first codec and the second codec can both be associated with a common payload format. The network gateway can modify first packets from a first to a second terminal to replace first-codec headers with second-codec headers and vice-versa.

A management device including an acquirer configured to acquire response information indicating whether a call-destination terminal has accepted participation in group communication, and terminal operation information indicating whether the call-destination terminal is in a state capable of receiving the group communication, a processor configured to create communication result information including participation information on the call-destination terminal whether to participate in the group communication, and a transmitter configured to transmit the communication result information, wherein, in a case of non-participation where the call-destination terminal has not participated in the group communication, the processor is configured to create the participation information that makes it possible to identify at least two factors of the non-participation.

Disclosed are a user terminal device and a control method therefor. The control method for a user terminal device comprises the steps of: determining state information of a satellite device connected with a user terminal device when a call request message is received from another terminal device; determining communication configuration information with the another terminal device on the basis of the state information of the satellite device, and transmitting a call response message including the communication configuration information to the another terminal device; and receiving audio data from the another terminal device according to a communication scheme corresponding to the communication configuration information. Therefore, in comparison with the prior art, the present invention can reduce a call negotiation time between user terminal devices, and thus reduce a call cost.

A network node, a communication device and methods therein, for handling dynamic uplink/downlink, UL/DL, subframe configurations when operating in Time Division Duplex, TDD. The network node obtains capability information indicating whether the communication device supports carrier aggregation and/or frequency band combination, and also indicating whether the communication device supports simultaneous reception and transmission of signals on different carriers or frequency bands. The network node then determines UL/DL subframe configurations for the communication device based on the obtained capability information such that the communication device is to use different UL/DL subframe configurations for different carriers or frequency bands, when certain conditions are fulfilled.

Interfacing according to a common public radio interface in a base station in a mobile communication system is described. The interfacing comprises a conversion process for rate-converting legacy data samples. First a predetermined number of the legacy data samples is converted to frequency samples in a frequency domain, then the frequency samples are zero padded to extend the frequency range according to a related sample rate of a 4G data format and then converted into a number of data samples of the related sample rate. The related sample rate is a multiplication of S/K times a basic frame rate of the 4G data format, S samples being allocated to K frames, K and S being integers and K being 8 or less. Advantageously large buffers for allocating a large number of legacy samples to 4G frames are avoided.

Speech transcoding in packet networks may be useful when both incoming and outgoing speech streams of the transcoding entity are packet based. This can be any transcoding entity having packet interfaces. A method can include omitting jitter buffering before decoding in a transcoder and omitting bad frame handling in a decoding stage of a transcoder. The method can also include freezing a decoder and the encoder when a packet is not received. The method can also include sending packet loss information from the decoder to the encoder as side information when the packet is not received. The method can further include setting an outgoing packet stream to permit detection of missing packets by a downstream decoder upon receiving a valid packet after the packet is not received.

The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.