A method for providing streaming audio over a network from a client to a server associated with a third-party service. The method includes receiving configuration data from a tenant regarding the server. The method may further include establishing a connection with the server via a connection process. The connection process includes sending a connection request that includes: a bidirectional communication protocol configured to transmit data from the client to the server and from the server to the client by reusing an established connection channel; a tenant-Id parameter; and a session-Id parameter. The method may further include implementing an open transaction for initiating the streaming of the audio of the ongoing conversation to the server. The open transaction includes transmitting participant parameters identifying the customer and at least one choice regarding an audio format for the audio streaming.

An exemplary method involves a user equipment (UE) receiving a communication-origination request to initiate a communication with a given counterpart. In response, the UE: (a) uses historical communication data for one or more previous communications as a basis for evaluating whether the UE is likely to be assigned a particular codec, from a plurality of possible codecs, for communication with the given counterpart; (b) based on results of the evaluation, determines a respective setting for each of one or more access parameters affecting a likelihood of successfully establishing the communication with the given counterpart; and (c) performs the communication-origination process according to the determined settings for the one or more access parameters, to initiate the requested communication with the given counterpart.

Provided are a terminal device and method of performing a call function transmitting ambient audio with high sensitivity.

A terminal device performing a call function with at least one external device via a network may include a receiver configured to receive at least one of an audio transmission signal and a video transmission signal to be transmitted to the external device; a processor configured to analyze at least one of the audio transmission signal and the video transmission signal, select one of a speech mode and an audio mode, based on a result of the analysis, and compress the audio transmission signal, based on the selected mode; a communicator configured to transmit the compressed audio transmission signal to the external device, and receive an audio reception signal from the external device; and an output unit configured to output the audio reception signal.

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.

According to one example, a method includes processing a communication session with a first virtual machine of a plurality of virtual machines associated with a network node and monitoring packet loss on a leg of the communication session between a first endpoint and a second endpoint. The method further includes, in response to determining that the packet loss exceeds a first threshold, toggling on an enhanced mode for a codec associated with the communication session, the enhanced mode providing increased error resilience. The method further includes, in response to determining that the toggling on the enhanced mode causes the first virtual machine to exceed a processing capacity threshold, moving the communication session to a second virtual machine of the plurality of virtual machines.

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.

A conference call management method, system, and computer program product include inferring an Internet Protocol (IP) address of a new user requesting to join a call including at least one other user, inferring a codec to stream an emulated network pattern for the call if the new user were to join the call, measuring a call quality perceived by the at least one other user in the call while the emulated codec is run on the call, and measuring an impact on the call quality in a case that the new user joins the call based on the perceived call quality and the network pattern.

Systems and methods are provided for operating an intercom system using a wireless access point. A codec is selected (501) from a plurality of available codecs. In some implementations, the available codecs present a tradeoff between audio quality and intercom device capacity. The access point operates (505) using the selected codec and, in response to detecting a new intercom device connecting to the intercom system (507) through the access point, the access point transmits a signal to the new intercom device identifying the selected codec. In response to a determination that the new intercom device does not have the selected codec stored in its memory, the access point automatically uploads (515) the codec to the new intercom device and transmits communications with the intercom deviceincluding, for example, audio stream datausing the selected codec.

A method, device and medium for determining a coding format are provided. The method includes: receiving one or more data packets forwarded by a call center during a VoLTE communication, in which the one or more data packets carry a first coding format; detecting whether the first coding format is same with a negotiated second coding format; and modifying the coding format used during the VoLTE communication from the second coding format to the first coding format, if the first coding format is not same with the negotiated second coding format.

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.