Embodiments provide systems, methods, apparatus, and computer program products for selecting codec parameters to satisfy one or more operating constraints. An example method performed by a network component that facilitates a communication session set-up process among endpoints in a communication network, the method includes: during the communication session set-up process, determining a utilization factor of the network component; selecting a value for sampling frequency associated with a first codec in response to determining the utilization factor; negotiating use of the first codec and the value for sampling frequency with a first endpoint and negotiating use of a second codec with a second endpoint; and transcoding a media stream of a communication session between the first endpoint and the second endpoint according to the first codec and the value for sampling frequency and the second codec.

Methods and systems for registering a phone device on a synthetic communication network are described. One implementation receives a request to connect the phone device to a synthetic communication network that transmits messages over a data network using one or more phone numbers from a voice network. A phone number and a country code associated with the phone device to be used for communications over the synthetic network are obtained. The phone number is normalized to a standard format distinguishing the resulting normalized phone number from international phone numbers using the synthetic network. Before the phone device is registered, an association between the normalized phone number and the phone device is verified by sending a verification message addressed to the phone device that loops back to the phone device. If the phone device confirms the verification as authentic, an authorization allows the phone device to communicate over the synthetic network.

A communication system is provided that includes a network node, a first terminal which supports an EVS (Enhanced Voice Service) codec, and a second terminal. The EVS codec supports an AMR-WB (Adaptive Multi Rate Wideband) codec compatible mode and an AMR-WB codec non-compatible mode, the AMR-WB codec having a dedicated payload format, where the AMR-WB codec compatible mode is compatible with an AMR-WB codec and the AMR-WB codec non-compatible mode is non-compatible with the AMR-WB codec. The first terminal and the second terminal negotiate the AMR-WB codec non-compatible mode in a session negotiation to select the AMR-WB codec non-compatible mode to be used for the communication between the first terminal and the second terminal when the communication starts.

Example configuration methods and apparatus are described. In one example method, a first baseband unit (BBU) updates a first line rate, and sends a data frame to the second BBU at an updated first line rate after each update. The first BBU receives a data frame that is sent by the second BBU at an updated second line rate after each time the second BBU updates a second line rate. When the first line rate is equal to the second line rate, the first BBU sends networking relationship information of the first BBU to the second BBU at the first line rate, and receives networking relationship information of the second BBU that is sent by the second BBU at the second line rate. The first BBU establishes a bidirectional upper-layer communication channel with the second BBU based on a communication address of the second BBU allocated by the first BBU.

A communication system is provided that includes a network node, a first terminal which supports an EVS (Enhanced Voice Service) codec, and a second terminal. The EVS codec supports an AMR-WB (Adaptive Multi Rate Wideband) codec compatible mode and an AMR-WB codec non-compatible mode, the AMR-WB codec having a dedicated payload format, where the AMR-WB codec compatible mode is compatible with an AMR-WB codec and the AMR-WB codec non-compatible mode is non-compatible with the AMR-WB codec. The first terminal and the second terminal negotiate the AMR-WB codec non-compatible mode in a session negotiation to select the AMR-WB codec non-compatible mode to be used for the communication between the first terminal and the second terminal when the communication starts.

Aspects of the present disclosure provide techniques and apparatus for wireless communication, and more particularly, increasing network coverage using rate adaption. The techniques include adjusting a cell transition threshold based on, for example, maximum packet loss rates (PLRs) for a VOIP terminal, for example, taking into consideration application layer redundancy employed at the terminal. Numerous other aspects are provided.

There is provided mechanisms for controlling media transcoding of a media session between at least two media transceiver devices. The method is performed by a network node. The method comprises obtaining, from at least one of the media transceiver devices, an indication of current allocation of energy for performing playback of the media session in the at least one of the media transceiver devices. The method comprises controlling transcoding of the media session according to the current allocation of energy of the at least one media transceiver device. The transcoding comprises either conversion from a first media format to a second media format of same media type, or conversion from a first media type to a second media type.

A network node is provided that communicates with a first terminal, which supports a first codec and a second terminal. The network node includes a detector that detects switching of a codec of the first terminal and the second terminal, wherein the first codec supports a codec A compatible mode and a codec A non-compatible mode, the first codec being a new codec and codec A having a dedicated payload format, and wherein the codec A compatible mode is compatible with a second codec and the codec A non-compatible mode is non-compatible with the second codec, the second codec being a legacy codec. The network node also includes a transmitter that transmits a signal for the second terminal to switch the negotiated codec A non-compatible mode of the second terminal to the codec A compatible mode.

Technologies for network configuration using dynamic voice codec and feature offering are described herein. A translation codec is used to allow a device capable of using enhanced voice services to use the enhanced voice services when communicating with a device that is not capable of using enhanced voice services.

The disclosure relates to methods and devices for controlling speech quality, in particular by controlling end-to-end latency and by improving speech quality in case of mobility scenarios. A method 200 for controlling end-to-end latency between receiving and processing audio frames includes: receiving 201 a data packet comprising at least one coded audio frame; storing 202 the received data packet in a packet buffer; retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame into audio samples; and processing 204 the audio samples, wherein a scheduling of retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame is based on a target criterion with respect to audio quality of the audio samples and latency between receiving the data packet and processing the audio samples, and wherein the scheduling is dynamically and smoothly shifted in time in order to avoid audio distortions.