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.

In some implementations, an application server may receive, from a calling party user equipment, a call for a called party associated with multiple user equipment. The application server may provide to the multiple user equipment, and based on the call, a request for transcoding information associated with the multiple user equipment. The application server may assign a transcoding resource for handling the call, wherein the transcoding resource is provided in a network. The application server may receive, based on the request, the transcoding information from a particular user equipment of the multiple user equipment. The application server may provide the transcoding information to the transcoding resource, wherein the transcoding information causes the transcoding resource to establish and transcode the call between the calling party user equipment and the particular user equipment.

A method and system for establishing optimized data streams in a network can be configured for crafting resource-draining Session Description Protocol (SDP) bodies on a call queue.

Various embodiments include methods for determining when to resynchronize voice-over IP (VoIP) communications of the wireless device. Various embodiments may include storing one or more first call characteristics of the VoIP communications between the wireless device and a first base station, detecting whether the VoIP communications are transferred from the first base station to a second base station, analyzing one or more second call characteristics of the VoIP communications between the wireless device and the second base station, determining whether the one or more second call characteristics differ from the one or more first call characteristics, determining, within a total wait time, whether no active voice frames will be transmitted across an uplink, or no active voice frames will be provided to audio decoding, resynchronizing the VoIP communications in response to determining that the one or more second call characteristics differ from the one or more first call characteristics.

Technology for transcoding avoidance is disclosed. A mobile switching center (MSC) server can decode a single radio voice call continuity (SRVCC) packet switch (PS) to circuit switched (CS) request message received from a mobility management entity (MME) that includes selected CODEC information for a selected CODEC used for a user equipment (UE) in an internet protocol (IP) Multimedia Subsystem (IMS) over long term evolution (LTE) system. The MSC server can encode the selected CODEC information for transmission to a target MSC to enable the target MSC to identify the selected CODEC for the UE to allow the selected CODEC to be used in the CS domain.

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.

Described herein are techniques, devices, and systems for providing an optimal voice experience over varying radio frequency (RF) conditions while using EVS audio codecs. A user equipment (UE) may adaptively transition between using a music-capable EVS codec (e.g., EVS-FB) as a default audio codec that provides a first audio bandwidth and a different EVS audio codec that provides a second audio bandwidth that is less than the first audio bandwidth. The transition to the different EVS audio codec may occur in response to determining a value indicative of a RF condition associated with a serving base station is less than a threshold value, which allows for providing preserving at least a minimal level of voice quality in degraded RF conditions.

A communication method supports Enhanced Voice Services (EVS) codec performed by a communication terminal apparatus. The method includes performing negotiation to use an EVS codec for communication between a communication terminal apparatus and a counterpart terminal, using an IP multimedia subsystem (IMS) signaling including one of a session description protocol (SDP) offer and an SDP answer, and performing negotiation to specify one or more audio-bandwidths of input signals in Hertz (Hz) or kilohertz (kHz) for the EVS codec. In a communication session after the codec negotiation session, the processor causes the receiver to receive a signaling for changing an audio-bandwidth of an input signal to the EVS codec from a network node, changes the audio-bandwidth of the input signal to the EVS codec to another audio-bandwidth without changing the EVS codec based on the signaling, and causes the transmitter to transmit encoded data in the changed audio-bandwidth.

A videoconferencing system includes a codec configured to receive a videoconferencing call, the codec including at least one application programming interface (API), a system controller configured to communicate with the codec using the at least one API, and at least one other videoconferencing component. The system controller is configured to send one or more commands to the at least one other videoconferencing component independently of the codec.

Described herein are techniques, devices, and systems for providing an optimal voice experience over varying radio frequency (RF) conditions while using EVS audio codecs. A user equipment (UE) may adaptively transition between using a music-capable EVS codec (e.g., EVS-FB) as a default audio codec that provides a first audio bandwidth and a different EVS audio codec that provides a second audio bandwidth that is less than the first audio bandwidth. The transition to the different EVS audio codec may occur in response to determining a value indicative of a RF condition associated with a serving base station is less than a threshold value, which allows for providing preserving at least a minimal level of voice quality in degraded RF conditions.