A network device determines radio frequency (RF) conditions at a first endpoint and a second endpoint of a call or session. The network device determines optimum first codec parameters for the determined RF conditions at the first endpoint of the call or session, and determines optimum second codec parameters for the determined RF conditions at the second endpoint of the call or session. The network device sends the first codec parameters to the first endpoint for altering operation of a first codec at a first device at the first endpoint. The network device sends the second codec parameters to the second endpoint for altering operation of a second codec at a second device at the second endpoint.

A communication terminal apparatus is provided that includes a negotiator that negotiates a codec used for communication between the communication terminal apparatus and a counterpart terminal at a start of the communication. The negotiator uses an IP multimedia subsystem (IMS) signaling including one of a session description protocol (SDP) offer and an SDP answer. The negotiated codec supports bandwidths of input signals of a plurality of codecs, and changes a bandwidth of an input signal during the communication with the counterpart terminal by changing a frequency range of the input signal. The apparatus also includes a bandwidth determiner that limits the bandwidth and an encoding bit rate of the input signal of the codec during the communication with the counterpart terminal, according to signaling for limiting the bandwidth and the encoding bit rate of the input signal of the codec, the signaling being notified from a network node.

Bandwidth-efficient (i.e., compressed) representations of audio data can be utilized for near real-time presentation of the audio on one or more receiving devices. Persons identified as having speech represented in the audio data can have trained speech models provided to the devices. These trained models can be used to classify the compressed audio in order to improve the quality to correspond more closely to the uncompressed version, without experiencing lag that might otherwise be associated with transmission of the uncompressed audio. The uncompressed audio is also received, with potential lag, and is used to further train the speech models in near real time. The ability to utilize the uncompressed audio as it is received prevents a need to store or further transmit the audio data for offline processing, and enables the further trained model to be used during the communication session.

In one example, the present disclosure describes a device, computer-readable medium, and method for improving media quality at the network edge by encoding the media into a digital stream that is carried, end-to-end, to a peer edge device. For instance, in one example, a method includes initiating, by a first computing device, a connection to a second computing device, selecting, by the first computing device, a codec for encoding data into a data stream, wherein the codec is selected such that the data stream can be decoded by the second computing device without being transcoded by an intermediary, encoding, by the first computing device, the data into the data stream using the codec, and sending, by the first computing device, the data stream to the second computing device.

A VoLTE communication method and a base station thereof are disclosed. The method includes: receiving a VoLTE communication request sent by a calling terminal; determining and sending a first suggestion complying with the VoLTE communication carried out by the calling terminal to a network side, the first suggestion including a first suggested speech coding mode and a corresponding code rate thereof; receiving from the network side a final speech coding mode and a corresponding code rate that are determined based on the first suggested speech coding mode and the corresponding code rate; and sending the final speech coding mode and the corresponding code rate to the calling terminal, such that the speech coding mode and the corresponding cod rate may be adjusted synchronously on the both side of the VoLTE communication.

Controlled-environment communication systems are increasingly using voice over internet protocol (VoIP) to serve their users. VoIP allows voice to be sent in packetized form, where audio is encoded using one of several codecs. Because of bandwidth constraints, particularly during peak call times, codecs may be used which sacrifice audio quality for bandwidth efficiency. As a result, several features of communication systems, including critical security features. The present disclosure provides details for systems and methods by which a controlled-environment communication system may shift between codecs to perform security-related features or to alleviate bandwidth considerations. This involves the special formatting of control-signaling messages, including session initiation protocol (SIP) and session description protocol (SDP) messaging.

A network device determines radio frequency (RF) conditions at a first endpoint and a second endpoint of a call or session. The network device determines optimum first codec parameters for the determined RF conditions at the first endpoint of the call or session, and determines optimum second codec parameters for the determined RF conditions at the second endpoint of the call or session. The network device sends the first codec parameters to the first endpoint for altering operation of a first codec at a first device at the first endpoint. The network device sends the second codec parameters to the second endpoint for altering operation of a second codec at a second device at the second endpoint.

Aspects of the subject disclosure may include, for example, including a processing system for performing operations for determining service requirements of a call session at first user equipment associated with a communication network, determining a first codec to facilitate the call session at the first user equipment according to the service requirements of the call session, searching a session border controller table according to the first codec to obtain a first resource identifier associated with a first session border controller type to facilitate the call session at the user equipment, receiving a first address of a first session border controller associated with the communication network from a domain name server associated with the communication network responsive to a first query including the first resource identifier, and sending a first transport protocol message to the first session border controller according to the first address. Other embodiments are disclosed.

A method is disclosed for wireless density-based on-the-fly codec and radio frequency link type selection. The method includes storing a link quality value, and scanning a plurality of timeslots across a plurality of wireless carriers. The method also includes calculating, based on the scanning, an average radio frequency density level for a time period. Still yet, the method includes calculating a current density level based on the link quality value and the average radio frequency density level. Further, the method includes selecting, based on the current density level, a codec from a plurality of stored codecs. The selected codec is associated with a link type. Moreover, the method includes receiving a command to open a wireless link, and, in response to the command, opening the wireless link using the selected codec and the associated link type.

In one example, the present disclosure describes a device, computer-readable medium, and method for improving media quality at the network edge by encoding the media into a digital stream that is carried, end-to-end, to a peer edge device. For instance, in one example, a method includes initiating, by a first computing device, a connection to a second computing device, selecting, by the first computing device, a codec for encoding data into a data stream, wherein the codec is selected such that the data stream can be decoded by the second computing device without being transcoded by an intermediary, encoding, by the first computing device, the data into the data stream using the codec, and sending, by the first computing device, the data stream to the second computing device.