An apparatus and mechanism to manage multi-modal real-time or quasi real-time sessions such as voice, video, instant messaging, or web-conferencing across multiple devices through a simple subscription to a session management service. A user can subscribe to the service from many devices (e.g. mobile device, desk phone, personal computer, internet appliance, internet TV, communication terminal, etc.) and through the service seamlessly move sessions back and forth between subscribed devices. In a preferred embodiment, a user utilizes a “virtual slider” in a simple “two touch” procedure to effectuate the movement of a communication session from one device to another device via the service.

A system for managing a conference call, the system comprising a call answering server configured to receive a call from an attendee device that has been forwarded from a host device, identify a conference call associated with the host device for the attendee device to join, receive a request from the attendee device to join the conference call, and connect the attendee device to the conference call.

This disclosure describes, in part, techniques for establishing network-based data communications (e.g., voice calls, video calls, etc.) between a user device of a user and a remote device of another user, and transitioning the data communications to a different user device of the user based on various types of information. In some examples, the user devices may be located in one or more environments of the user, and the data communications may be transitioned between the user devices based, at least in part, on a location of the user in the environment(s) relative to the multiple devices. For instance, if a user device is performing data communications with the remote device, but it is determined that the user has moved into a closer proximity to another user device, the performance of the data communications may be transitioned to the other user device to which the user is in closer proximity.

A system controls the audio focus for a user in a virtual conference call. The system retrieves sound parameters associated with participants in the virtual conference call with the user from a user profile of the user. The sound parameters define volume adjustments to be applied to audio data received from the participants for generating an audio mix customized to the user. The system receives the audio data from client devices of the participants, and for each of the participants, adjusts the audio data of the participant using the associated sound parameter of the participant. The system adds the adjusted audio data of the participants to the audio mix for the user and provides the audio mix to a client device of the user.

In a teleconferencing method, a first media stream and a second media stream of a teleconference are received, by processing circuitry of a first device, from a second device. The first media stream includes first audio and the second media stream includes second audio. Default weight information is received from the second device. The default weight information indicates a first audio weight for weighting the first audio and a second audio weight for weighting the second audio. The first audio weight for weighting the first audio and the second audio weight for weighting the second audio are determined based on the default weight information. Mixed audio is generated, by the processing circuitry of the first device, by combining a weighted first audio based on the first audio weight applied to the first audio and a weighted second audio based on the second audio weight applied to the second audio.

A call processing method and an electronic device are provided. The method is performed by an electronic device and includes: in a case that a call is connected successfully, obtaining the number of speakers in the current call. The method further includes: when the number of speakers is greater than a preset threshold, determining whether a user of the electronic device is a target user. The method additionally includes: if the user of the electronic device is a non-target user, disabling a speaking channel of the electronic device; and if the user of the electronic device is a target user, keeping the speaking channel of the electronic device on.

Systems and methods for providing headset/ear bud voice control to crew members in a quick-service restaurant are disclosed. Exemplary implementations may: transmit transmission signals from a headset to the base station, receive reception signals by the headset from the base station; capture audio information from a first crew member, generate sound based on the received reception signals, generate the transmission signals to form a headset-specific uplink packet based on the captured audio information; generate semantic information based on the headset-specific uplink packet, wherein the semantic information includes one or more individual words; determine one or more commands that modify operation of the communication system, based on the semantic information, and facilitate execution of the one or more commands to control and/or modify the operation of the communication system.

Systems and methods for providing headset/ear bud voice control to crew members in a quick-service restaurant are disclosed. Exemplary implementations may: transmit transmission signals from a headset to the base station, receive reception signals by the headset from the base station; capture audio information from a first crew member, generate sound based on the received reception signals, generate the transmission signals to form a headset-specific uplink packet based on the captured audio information; generate semantic information based on the headset-specific uplink packet, wherein the semantic information includes one or more individual words; determine one or more commands that modify operation of the communication system, based on the semantic information, and facilitate execution of the one or more commands to control and/or modify the operation of the communication system.

One or more embodiments of the present specification relates to an online interview method and system. The method includes: establishing communication connection between an interviewing terminal and an interviewed terminal through a network; and acquiring communication information between the interviewing terminal and the interviewed terminal. The communication information includes one or more types of audio information, video information, and text information. The interviewing terminal includes one or more of a first host terminal, a second host terminal, and a text processing terminal. The first host terminal is configured to host an interview, and the first host terminal displays an interview outline and/or information of the interviewed terminal. The second host terminal is configured to host the interview and/or participate in the consultation of interview questions. The text processing terminal converts the audio information and/or the audio information in the video into corresponding text information.

Embodiments described herein provide for the dynamic adjustment of traffic associated with audiovisual conferences or other types of communication sessions in situations where a network connection of one or more conference participants exhibits issues that may affect audio and/or video quality. The adjustment may include the determination at a particular conference participant (e.g., a particular User Equipment (“UE”)) of degraded network conditions, the generation of condensed conference information at the UE, and the outputting of the condensed conference information via the network. The condensed conference information may be included in control signaling. The condensed conference information may be used to generate reconstructed conference information, which may be distributed to other conference participants.