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.

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 device for transmitting a media stream over a network, the device comprising: a quality estimator configured to maintain a measure of network quality indicative of available bandwidth over the network; a configuration manager configured to: use the measure of network quality to detect a reduction in network quality indicative of a reduced available bandwidth and, responsive to detecting said reduction, estimate a first bitrate corresponding to the reduced available bandwidth; in response to estimating the first bitrate, determine a second bitrate lower than the first bitrate; and a transceiver configured to transmit the media stream at a transmission bitrate substantially equal to the second bitrate for a first period of time, wherein the configuration manager is further configured to cause an increase of the transmission bitrate from the second bitrate to the first bitrate over a second period of time.

Various of the disclosed embodiments improve encoding during a network call, such as a Voice Over Internet Protocol (VOIP) connection, by adjusting the size of a data communications packet (“packet”). Particularly, given a corpus of codecs with which to encode data, the embodiments may identify a packet size based upon a common multiple of each codec's minimum raw data size. The packet size may be selected to accommodate the inclusion of data encoded in each codec format, as well as error correction code data, and codec transition commands. The packet size may be tailored to trade off measured latency and data efficiency.

In various embodiments, a corpus of codecs may be correlated with different, partially overlapping ranges of transmission characteristics. As channel conditions degrade or improve, the system may select a new codec with which to continue the connection based upon the corresponding overlapping range. Codecs may not be switched immediately when the transmission characteristics enter overlapping ranges, to avoid degrading the user's experience. If the characteristics remain in the overlap, or manifest a likely progression toward another region, then the transition may be effected.

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.

Aspects of the subject disclosure may include, for example, providing a request associated with a call session to a server, wherein the request includes capability information associated with user equipment. A codec is identified according to the capability information to obtain an identified codec, which facilitates media service to the user equipment. The media service is accessed from a media resource function, which in turn, is accessed by the server responsive to the identified codec. An operable resource identifier is identified responsive to a search of a codec table, wherein the server accesses the operable resource identifier associated with the media resource function from a domain name server responsive to the resource identifier not being available at the codec table. Other embodiments are disclosed.

Concepts and technologies provided herein can provide high-definition voice for fixed mobile convergence on mobile and wireline networks. A processor executing instructions can detect a call request associated with a called telephone number to setup a call session, where the call request is initiated from a calling device associated with an originating network. The processor can determine a call path for the call session from the originating network to a receiving network. The processor can create a fixed mobile convergence request to alert an electronic number mapping system of the call path for the call session from the originating network to the receiving network. The electronic number mapping system can provide a network presence map identifying a plurality of call receiving devices associated with the called telephone number that are available to participate in the call session via the receiving network.

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.

Connecting agent to customers is the cornerstone of contact center operations. When a customer is connected to an agent for real-time communications (e.g., voice), they can be particular sensitive to any delay, such as the delay to establish a media channel between the agent's endpoint and the customer's device. By establishing a media channel between the agent's device and a server having initially received the communication, the communication may be attached to the existing media channel without requiring the time and resources otherwise necessary to establish the media channel. Additionally, if a problem or failure occurs, a server utilized for the connection may seek to establish the media channel with a second network, such as a telephone network, or a second agent endpoint or endpoint component.