Provided is a method for detecting an audio input interface, comprising: acquiring input signals of every one of audio input interfaces; acquiring energy values of said input signals of every one of said audio input interfaces, and acquiring an interface identification of the audio input interface of which the energy value is the maximum among the energy values; adding said interface identification acquired herein into an identification sequence in an order by acquiring time; and identifying the audio input interface, of which the interface identification is saved most often in said identification sequence, as a valid audio input interface. Also provided is a system for detecting an audio input interface. The present disclosure is capable of effectively identifying the audio input interface connected to the microphone that the user is using, so that the user does not need to switch manually.

A system and method for providing radio access network (RAN) aware service delivery for Push-to-talk-over-Cellular (PoC) networks, wherein at least one of the servers performs an presence delivery method that reduces presence notifications delivered to the mobile units participating in communications services. The effective presence delivery method may comprise: throttling the delivery of the presence notifications; avoiding simultaneous dispatch of the presence notifications to multiple watchers located in the same cell or group of cells of the wireless communications network; or the server delaying sending the presence notifications for a mobile unit until the mobile unit triggers the server.

Meeting and conferencing systems and methods are implemented in a variety of manners. Consistent with one example embodiment, an audio conference system includes communications circuitry and a caller interface, each associated with setting up or managing the audio conference. Menu options may be provided to validate or authenticate incoming telephony calls. In response to a selection of the menu options, an incoming telephony call may be prompted to select a callback number, and then the system monitors for a trigger event. A participant is contacted in response to detecting the trigger event, and the contacted participant is then placed into the audio conference.

A method performed by a computing system includes managing an audio communication session that includes an audio connection between a plurality of participant devices, associating with each of the plurality of participant devices, a volume adjustment factor, adjusting a volume of an audio out signal from a first participant device of the plurality of participant devices to create an adjusted audio out signal, the adjusting being based on the volume adjustment factor associated with the first participant device, and transmitting a conference audio signal based on the adjusted audio out signal to a subset of the plurality of participant devices.

A system, method, and computer readable medium that facilitate diagnosing a conference call are provided. A tap is deployed at the ingress side and the egress side of one or more processing systems that handle each leg of a conference. When a conference is disrupted by noise, loss, or the like, an operator may initiate a command to systemically record each leg of the conference for a predetermined period of time. Each leg may then be analyzed to determine if it is the source of the disruption.

The present document relates to audio conference systems. In particular, the present document relates to improving the perceptual continuity within an audio conference system. According to an aspect, a method for multiplexing first and second continuous input audio signals is described, to yield a multiplexed output audio signal which is to be rendered to a listener. The first and second input audio signals (123) are indicative of sounds captured by a first and a second endpoint (120, 170), respectively. The method comprises determining a talk activity (201, 202) in the first and second input audio signals (123), respectively; and determining the multiplexed output audio signal based on the first and/or second input audio signals (123) and subject to one or more multiplexing conditions. The one or more multiplexing conditions comprise: at a time instant, when there is talk activity (201) in the first input audio signal (123), determining the multiplexed output audio signal at least based on the first input audio signal (123); at a time instant, when there is talk activity (202) in the second input audio signal (123), determining the multiplexed output audio signal at least based on the second input audio signal (123); and at a silence time instant, when there is no talk activity (201, 202) in the first and in the second input audio signals (123), determining the multiplexed output audio signal based on only one of the first and second input audio signals (123).

The present document relates to audio conference systems. In particular, the present document relates to the mapping of soundfields within an audio conference system. A conference multiplexer (110, 175, 210, 400) configured to place a first input soundfield signal (402) originating from a first soundfield endpoint (120, 170) within a 2D or 3D conference scene (300) to be rendered to a listener (301) is described. The first input soundfield signal (402) is indicative of a soundfield captured by the first soundfield endpoint (120, 170). The conference multiplexer (110, 175, 210, 400) is configured to set up the conference scene (300) comprising a plurality of talker locations (321, 322, 332, 331) at different angles (323, 333) with respect to the listener (301); provide a first sector (325); wherein the first sector (325) has a first angular width (324); wherein the first angular width (324) is greater than zero; and transform the first input soundfield signal (402) into a first output soundfield signal (403), such that for the listener (301) the first output soundfield signal (403) appears to be emanating from one or more virtual talker locations (321, 322) within the first sector (325).

A method for creating, modifying, and terminating connections between Internet end systems, particularly, although not exclusively, for Internet telephony communication. The method relies on several request messages sent between a client and a server and the response messages sent back in response. Each request and response message may contain one or more header fields which modify or more uniquely link the messages with a given connection. On this basis, advanced telephony services, such as call forwarding, call transferring, and multiparty conferencing are provided.