Abstract: Teleconferencing systems may adapt to changes associated with participating terminals by monitoring characteristics of the terminals and modifying audio streams in accordance with detected changes. A terminal moving from a location shared with one or more other participating terminals may be detected in a variety of ways. In response to the location change, a teleconference server may modify the audio stream transmitted to the terminal so that audio originating from the one or more other participating terminals is included in the stream. Additionally, audio streams transmitted to the one or more other terminals may also be modified to include audio signals originating from the moved terminal. Other changes to a terminal that may also affect audio processing include activating a muting function. Muting a terminal while the terminal is among a group of participating terminals may cause the audio signals from the other terminals to also be muted.

Abstract: An improved application programming interface (API) as described can control media resources in numerous Internet multimedia applications. The API may be independent of the application itself and the media resources underneath. The API may be referred to as a multimedia subsystem (MSS) interface.

Abstract: It is disclosed a method comprising registering a respective target network system state from each one of a plurality of programs, each of the plurality of programs comprising a first program state, detecting a current network system state, selecting one or more of the plurality of programs based on a result matching the detected current network system state against the registered target network system states, and commanding transition of the selected one or more programs from the first program state to a second program state different from the first program state.

Abstract: It is disclosed a method comprising registering a respective target network system state from each one of a plurality of programs, each of the plurality of programs comprising a first program state, detecting a current network system state, selecting one or more of the plurality of programs based on a result matching the detected current network system state against the registered target network system states, and commanding transition of the selected one or more programs from the first program state to a second program state different from the first program state.

Abstract: Teleconferencing systems may adapt to changes associated with participating terminals by monitoring characteristics of the terminals and modifying audio streams in accordance with detected changes. A terminal moving from a location shared with one or more other participating terminals may be detected in a variety of ways. In response to the location change, a teleconference server may modify the audio stream transmitted to the terminal so that audio originating from the one or more other participating terminals is included in the stream. Additionally, audio streams transmitted to the one or more other terminals may also be modified to include audio signals originating from the moved terminal. Other changes to a terminal that may also affect audio processing include activating a muting function. Muting a terminal while the terminal is among a group of participating terminals may cause the audio signals from the other terminals to also be muted.

Abstract: Circuitry, embodied in a media subsystem (10A), reproduces a speech or other type of audio signal, and is operable when playing back audio data for reducing synchronization delay. A method operates by, when a frame containing audio data is sent to a decoder (20), measuring the synchronization delay; determining by how much the synchronization delay should be adjusted; and adjusting the synchronization delay in a content-aware manner by adding or removing one or more audio samples in a selected current frame or in a selected subsequent frame so as not to significantly degrade the quality of the played back audio data. When the synchronization delay is adjusted by more than one audio sample, the adjustment can be made by all of the determined audio samples in one adjustment, or the adjustment is made by less than all of the determined audio samples by a plurality of adjustments. The step of adjusting selects, if possible, an unvoiced frame and discriminates against a transient frame.

Abstract: A system and method for applying a SIP events framework in order to deliver syndication feeds. The invention includes a feed delivery server and a feed consumption client. The feed consumption client uses a SIP SUBSCRIBE method to subscribe to the feed. Once the feed consumption client has subscribed to the feed, the feed delivery server starts delivering new feed items using a SIP NOTIFY method whenever new information becomes available, providing a PUSH mechanism where new material is provided to the client when available.

Abstract: A system and method for providing mobile assisted, fixed line communication. A device connectable to a mobile network coordinates with a session gateway device that is connected to a fixed line network in order to establish a connection to another gateway device. Signaling over a cellular or other non-fixed line network is used to establish connection parameters relating to a fixed line session. The established connection parameters are transmitted to the respective session gateway devices. Based upon these parameters, the session gateway devices can directly communicate with each other over the Internet or other fixed line network.

Abstract: Circuitry, embodied in a media subsystem, reproduces a speech or other type of audio signal, and is operable during a time that comfort noise is being generated for storing data from an arriving data packet that contains data representing a beginning of an interval of speech. The circuitry detects that data representing speech has been stored and synchronizes the circuitry to the beginning of the interval of speech by terminating the generation of comfort noise that precedes the interval of speech, and reproduces the speech by decoding the stored data from the data packet and converting the decoded data to an audio speech signal. The arriving data packet contains at least one frame of speech data having a duration of X milliseconds, and the circuitry includes an audio device, such as a DAC or a PCM connection, that requests decoded speech data at a rate given by X/Y milliseconds, where Y is greater than one.

Abstract: The present invention relates to the exchange of images and image related data between at least two end points 10, 11, 12, 13, in which exchange at least one wireless terminal 10, 11, 12 is a participating end point. The invention is based on the insight to use the Session Initiation Protocol as an upper layer protocol to enable usage of a client-server protocol that handles images, such that efficient transfer of image related data between a client 10, 11, 12 and a server 10, 11, 12, 13 is achieved. An embodiment of the invention is based on the further insight to use the Session Initiation Protocol as an upper layer protocol to enable usage of the JPIP protocol, which latter is a client-server protocol for handling images, such that efficient transfer of image related data between a client and a server is achieved.

Abstract: There is described an improved application programming interface (API) to control media resources in numerous Internet multimedia applications. The API is independent of the application itself and the media resources underneath. For the purposes of description, the API is called a multimedia subsystem (MSS) interface.

Abstract: A system, apparatus, and method is disclosed for providing media session descriptors. In accordance with one embodiment of the invention, a method of initiating a multimedia session via a computing arrangement involves forming a request for a session descriptor usable for initiating the multimedia session. The request is communicated to a session descriptor module of the computing arrangement. A session descriptor is formed based on a set of multimedia session parameters of the computing arrangement. The session descriptor is communicated from the session descriptor module, and the multimedia session is initiated using the session descriptor.

Abstract: A method and system for text-to-speech (TTS) service in a network that includes forming a network address to a destination node in the network. Text is inserted into a field of the address. The address is received at the destination node. The text is converted to speech at the destination node. The speech is then sent to a node in the network.

Abstract: Circuitry, embodied in a media subsystem (10A), reproduces a speech or other type of audio signal, and is operable when playing back audio data for reducing synchronization delay. A method operates by, when a frame containing audio data is sent to a decoder (20), measuring the synchronization delay; determining by how much the synchronization delay should be adjusted; and adjusting the synchronization delay in a content-aware manner by adding or removing one or more audio samples in a selected current frame or in a selected subsequent frame so as not to significantly degrade the quality of the played back audio data. When the synchronization delay is adjusted by more than one audio sample, the adjustment can be made by all of the determined audio samples in one adjustment, or the adjustment is made by less than all of the determined audio samples by a plurality of adjustments. The step of adjusting selects, if possible, an unvoiced frame and discriminates against a transient frame.

Abstract: Circuitry, embodied in a media subsystem, reproduces a speech or other type of audio signal, and is operable during a time that comfort noise is being generated for storing data from an arriving data packet that contains data representing a beginning of an interval of speech. The circuitry detects that data representing speech has been stored and synchronizes the circuitry to the beginning of the interval of speech by terminating the generation of comfort noise that precedes the interval of speech, and reproduces the speech by decoding the stored data from the data packet and converting the decoded data to an audio speech signal. The arriving data packet contains at least one frame of speech data having a duration of X milliseconds, and the circuitry includes an audio device, such as a DAC or a PCM connection, that requests decoded speech data at a rate given by X/Y milliseconds, where Y is greater than one.