Patent classifications
H04M7/0072
High-Definition Voice for Fixed Mobile Convergence on Mobile and Wireline Networks
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.
Utilizing VoIP coded negotiation during a controlled environment call
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.
TRANSCODING AVOIDANCE DURING SINGLE RADIO VOICE CALL CONTINUITY (SRVCC)
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.
Presentation Systems And Related Methods
A videoconferencing system includes a codec configured to generate one or more acknowledgement signals each time a predefined action or event occurs, and a remote control in communication with the codec for controlling the codec and for receiving the one or more acknowledgement signals generated by the codec. The remote control includes a user interface for displaying information to a user. The remote control is configured to update the information displayed to the user in response to receiving the one or more acknowledgement signals generated by the codec.
METHOD FOR ACHIEVING REMOTE ACCESS TO A PERSONAL VOICE ASSISTANT
A method for achieving remote access to a voice assistant suitable for setting up telephone communications with a communication terminal via a telecommunications network. The method includes: setting up a first communication with a caller terminal; obtaining, via the set-up communication, a datum identifying the voice assistant; transmitting a setup message for setting up a second communication to the identified voice assistant, the message containing at least one parameter suitable for activating a remote operating mode of the voice assistant; and connecting the first and second communications.
METHOD AND APPARATUS FOR PROVIDING MEDIA RESOURCES IN A COMMUNICATION NETWORK
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.
METHODS FOR INCREASING VOICE-OVER-INTERNET PROTOCOL (VOIP) NETWORK COVERAGE
The disclosure generally relates to various methods to increase network coverage for a Voice-over-Internet Protocol (VoIP) session between a first user equipment (UE) and a second UE. In an aspect, a first and second UEs negotiate a codec configuration to use in the VoIP session, transmits, to the second UE, a maximum end-to-end packet loss rate (PLR) that the first UE can tolerate for received media given the negotiated codec configuration, receives, from the second UE, a maximum end-to-end PLR that the second UE can tolerate for received media given the negotiated codec configuration, and determines a distribution of the maximum end-to-end PLRs among respective uplinks and downlinks at the first UE and the second UE.
Terminal device and method for performing call function
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.
Toggling enhanced mode for a codec
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.
Adaptability in EVS codec to improve power efficiency
This disclosure relates to techniques for codec selection for voice calls. A wireless device may initiate a voice call and may determine one or more link quality indicators. Based at least in part on the one or more link quality indicators, the wireless device may select a codec for the voice call.