Abstract: A speech signal processing method of a user terminal includes: receiving a speech signal, detecting a personalized information section including personal information in the speech signal, performing data processing on the personalized information section of the speech signal by using a personalized model generated based on the personal information, and receiving, from a server, a result of the data processing performed by the server on a general information section of the speech signal that is different than the personalized information section of the speech signal.

Abstract: An audio signal coding apparatus includes a time-frequency transformer that outputs sub-band spectra from an input signal; a sub-band energy quantizer; a tonality calculator that analyzes tonality of the sub-band spectra; a bit allocator that selects a second sub-band on which quantization is performed by a second quantizer on the basis of the analysis result of the tonality and quantized sub-band energy, and determines a first number of bits to be allocated to a first sub-band on which quantization is performed by a first quantizer; the first quantizer that performs first coding using the first number of bits; the second quantizer that performs coding using a second coding method; and a multiplexer.

Abstract: Certain example embodiments relate to speech privacy systems and/or associated methods. The techniques described herein disrupt the intelligibility of the perceived speech by, for example, superimposing onto an original speech signal a masking replica of the original speech signal in which portions of it are smeared by a time delay and/or amplitude adjustment, with the time delays and/or amplitude adjustments oscillating over time. In certain example embodiments, smearing of the original signal may be generated in frequency ranges corresponding to formants, consonant sounds, phonemes, and/or other related or non-related information-carrying building blocks of speech. Additionally, or in the alternative, annoying reverberations particular to a room or area in low frequency ranges may be “cut out” of the replica signal, without increasing or substantially increasing perceived loudness.

Abstract: To provide an education support system that allows educators to conduct adequate assessments considering handicaps or characteristics of individual students and to offer effective and efficient learning.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: In one aspect, a first device includes at least one processor, a microphone accessible to the at least one processor, a communication interface accessible to the at least one processor, and storage accessible to the at least one processor. The storage bears instructions executable by the at least one processor to await first voice input comprising a response cue and receive, from the microphone, the first voice input comprising the response cue. The instructions are also executable by the at least one processor to, responsive to receipt of the first voice input comprising the response cue, transmit a wake up command to a second device different from the first device. The wake up command is transmitted via the communication interface using a peripheral device communication protocol.

Abstract: Technologies for privately processing voice data include a compute device configured to continually or periodically capture voice data of a user by the compute device. The captured voice data is processed to remove or reduce the user's privacy-sensitive information. For example, the compute device fragments the captured voice data to generate a set of voice data fragments and further scrambles the voice data fragments to generate scrambled voice data fragments having a sequential order different from the plurality of voice data fragments. To scramble the voice data fragments, the compute device may reorder the voice data fragments such that each fragment is repositioned from its corresponding original sequential position in the voice data by a particular number of words, syllables, or phrases.

Abstract: A processing unit can operate an end-to-end recurrent neural network (RNN) with limited contextual dialog memory that can be jointly trained by supervised signals-user slot tagging, intent prediction and/or system action prediction. The end-to-end RNN, or joint model has shown advantages over separate models for natural language understanding (NLU) and dialog management and can capture expressive feature representations beyond conventional aggregation of slot tags and intents, to mitigate effects of noisy output from NLU. The joint model can apply a supervised signal from system actions to refine the NLU model. By back-propagating errors associated with system action prediction to the NLU model, the joint model can use machine learning to predict user intent by a binary classification obtained by both forward and backward output, and perform slot tagging, and make system action predictions based on user input, e.g., utterances across a number of domains.

Abstract: Embodiments of the present application relate to a method for speech recognition processing, a system for speech recognition processing, and a computer program product for speech recognition processing. A method for speech recognition processing is provided. The method includes receiving a speech recognition request, the speech recognition request including a pre-assigned speech ID that uniquely identifies speech information and a speech recognition command to convert speech into text, locating speech information based on the speech ID, automatically converting the speech information to obtain text information corresponding to the speech information in response to the speech recognition command, and sending the obtained text information as a speech recognition result to a user that caused the speech recognition request to be issued.

Abstract: A method of automatic discovery of analysis scripts for a dataset, the method including: utilizing at least one processor to execute computer code that performs the steps of: receiving, at a script searching tool, an input dataset; searching, in a script repository, a plurality of datasets having analysis scripts associated therewith; the searching comprising finding, based on a feature of the input dataset, one or more datasets of the plurality of datasets having the feature; identifying, based on the one or more datasets of the plurality of datasets having the feature, one or more associated analysis scripts; and returning, via a user interface, a result listing of the one or more associated analysis scripts. Other aspects are described and claimed.

Abstract: The present application relates to packet loss concealment apparatus and method, and audio processing system. According to an embodiment, the packet loss concealment apparatus is provided for concealing packet losses in a stream of audio packets, each audio packet comprising at least one audio frame in transmission format comprising at least one monaural component and at least one spatial component. The packet loss concealment apparatus may comprises a first concealment unit for creating the at least one monaural component for a lost frame in a lost packet and a second concealment unit for creating the at least one spatial component for the lost frame. According to the embodiment, spatial artifacts such as incorrect angle and diffuseness may be avoided as far as possible in PLC for multi-channel spatial or sound field encoded audio signals.

Abstract: The MAC frame in a wireless communication system includes a terminal ID allocated to each of multiple terminals. At least one connection ID is allocated to each terminal having the terminal ID, and sub-carrier allocation information is allocated to each connection having the connection ID. The sub-carrier allocation information includes a sub-carrier allocation status for each sub-carrier, and the number of allocated information bits for each sub-carrier. The sub-carrier allocation status and the number of allocated information bits for each sub-carrier can be allocated, by sub-carriers, to the sub-carrier allocation information using a same number of bits; or the information on the sub-carrier allocation status is first allocated to the sub-carrier allocation information and the number of allocated information bits for each sub-carrier is allocated.

Abstract: A sampler module may divide an audio signal into a series of sequential samples. A signal quality detector module may identify a consistent brick wall frequency of the audio signal spanning a plurality of the sequential samples at an outset of the audio signal and determine a signal treatment indication proportional to the brick wall frequency. A signal enhancer module may sequentially receive and analyze one or more sample components of the audio signal to identify lost parts of the audio signal in the one or more sample components of respective sequential samples, and generate, in accordance with the signal quality indication, a corresponding signal treatment for each of the one or more sample components of respective sequential samples having a corresponding identified lost part.

Abstract: A method for removing transitional noise in an audio data stream is performed by a computing device. After obtaining first and second audio frames in a time sequence of the audio data stream, the computing device determines a frame segmentation position in the audio frames based on predefined closeness criteria that require a predefined distance between the sampling point values at the frame segmentation position in the first and second audio frames to satisfy a predefined closeness condition. Next, the computing device obtains respective sampling point values preceding the frame segmentation position in the second audio frame and ones following the frame segmentation position in the first audio frame. The computing device then sequentially stitches the respective sampling point values obtained from the first and second audio frame to generate a third audio frame and inserts the third audio frame between the first audio frame and the second audio frame.

Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for removing jitter introduced by a packet switched network. Each received audio frame comprises a primary portion and a redundancy portion. The redundancy portion comprises a partial redundant copy of a previous frame that is offset by k frames. If a frame n is lost, a frame n+k that comprises the partial redundant copy of the lost frame n, is located in a jitter buffer. Based on the frame n+k, a substitute frame n? substituting the lost frame n is created and a substitution indicator of the substitute frame n? is set to indicate that the redundancy portion of the substitute fame n? should be used in decoding.

Abstract: The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

Abstract: Embodiments of the present application relate to a method for speech recognition processing, a system for speech recognition processing, and a computer program product for speech recognition processing. A method for speech recognition processing is provided. The method includes receiving a speech recognition request, the speech recognition request including a pre-assigned speech ID that uniquely identifies speech information and a speech recognition command to convert speech into text, locating speech information based on the speech ID, automatically converting the speech information to obtain text information corresponding to the speech information in response to the speech recognition command, and sending the obtained text information as a speech recognition result to a user that caused the speech recognition request to be issued.

Abstract: A method for hybrid speech enhancement which employs parametric-coded enhancement (or blend of parametric-coded and waveform-coded enhancement) under some signal conditions and waveform-coded enhancement (or a different blend of parametric-coded and waveform-coded enhancement) under other signal conditions. Other aspects are methods for generating a bitstream indicative of an audio program including speech and other content, such that hybrid speech enhancement can be performed on the program, a decoder including a buffer which stores at least one segment of an encoded audio bitstream generated by any embodiment of the inventive method, and a system or device (e.g., an encoder or decoder) configured (e.g., programmed) to perform any embodiment of the inventive method. At least some of speech enhancement operations are performed by a recipient audio decoder with Mid/Side speech enhancement metadata generated by an upstream audio encoder.

Abstract: A parametric stereo upmix method for generating a left signal and a right signal from a mono downmix signal based on spatial parameters includes predicting a difference signal comprising a difference between the left signal and the right signal based on the mono downmix signal scaled with a prediction coefficient. The prediction coefficient is derived from the spatial parameters. The method further includes deriving the left signal and the right signal based on a sum and a difference of the mono downmix signal and said difference signal.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for adjusting acoustic parameters. In one aspect, a method includes receiving an identifier associated with an enclosure for a computing device, transmitting data identifying the identifier associated with the enclosure for the computing device, and receiving one or more physical parameters of the enclosure for the computing device. The method also includes based on the one or more physical parameters of the enclosure for the computing device, determining, one or more acoustic parameter adjustments of the computing device in the enclosure, the one or more acoustic parameter adjustments being configured to preserve one or more acoustic characteristics of the computing device out of the enclosure while the computing device is in the enclosure, and based on the one or more acoustic parameter adjustments, adjusting the one or more acoustic parameters of the computing device.

Abstract: The invention relates to a codec and a signal classifier and methods therein for signal classification and selection of a coding mode based on audio signal characteristics. A method embodiment to be performed by a decoder comprises, for a frame m: determining a stability value D(m) based on a difference, in a transform domain, between a range of a spectral envelope of frame m and a corresponding range of a spectral envelope of an adjacent frame m?1. Each such range comprises a set of quantized spectral envelope values related to the energy in spectral bands of a segment of the audio signal. The method further comprises selecting a decoding mode, out of a plurality of decoding modes, based on the stability value D(m); and applying the selected decoding mode.

Abstract: Provided are an apparatus and a method for integrally encoding and decoding a speech signal and an audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is an audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; a frequency band expander for expanding a frequency band of the input signal whose output is transmitted to either the time-domain encoding module or the transform encoding module based on the input characteristic; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.

Abstract: In one example, an edge routing device of a service provider network includes one or more network interfaces configured to send and receive packets and a processing unit configured to retrieve, from a packet received via the one or more network interfaces, priority data from an Internet protocol (IP) header of the packet, form a first tag including a first set of data in a first priority field of the first tag, wherein the first set of data represents a first portion of the priority data, form a second tag including a second set of data in a second priority field of the second tag, wherein the second set of data represents a second portion of the priority data, encapsulate the packet with the first tag and the second tag, and forward, via the one or more network interfaces, the encapsulated packet.

Abstract: The present invention relates to an audio encoding and, more particularly, to a signal classifying method and device, and an audio encoding method and device using the same, which can reduce a delay caused by an encoding mode switching while improving the quality of reconstructed sound. The signal classifying method may comprise the operations of: classifying a current frame into one of a speech signal and a music signal; determining, on the basis of a characteristic parameter obtained from multiple frames, whether a result of the classifying of the current frame includes an error; and correcting the result of the classifying of the current frame in accordance with a result of the determination. By correcting an initial classification result of an audio signal on the basis of a correction parameter, the present invention can determine an optimum coding mode for the characteristic of an audio signal and can prevent frequent coding mode switching between frames.

Abstract: Disclosed are techniques and systems to provide a narration of a text in multiple different voices. In some aspects, systems and methods described herein can include receiving a user-based selection of a first portion of words in a document where the document has a pre-associated first voice model and overwriting the association of the first voice model, by the one or more computers, with a second voice model for the first portion of words.

Abstract: Techniques are provided for a playback device to play a media item using an audio setting corresponding to the media item and characteristics of the playback device. An example implementation involves a first playback device transmitting, to a first computing device, information indicating one or more characteristics of the first playback device and queue information indicating one or more media items in a playback queue to be played by the first playback device. The example implementation may further involve receiving, from the first computing device, one or more audio settings associated with the one or more media items, the one or more audio settings corresponding to the one or more media items and the one or more characteristics of the first playback device and playing a first media item of the one or more media items according to a first audio setting of the one or more audio settings.

Abstract: Systems and methods for contacting emergency personnel via voice recognition are provided. Some methods include detecting and identifying, via the voice recognition, a first predetermined phrase, detemiining whether a second predetermined phrase is detected and identified, via the voice recognition, within a predetermined period of time after detection and identification of the first predetermined phrase, and when the second predetermined phrase is detected and identified within the predetermined period of time, contacting the emergency personnel.

Abstract: Disclosed are a method of controlling a source device in a sink device and apparatus therefor. The present invention includes the steps of outputting the video data received from the source device to a display unit, recognizing a text from a screen corresponding to the video data outputted to the display unit, saving the recognized text to a memory to correspond to location information on the screen corresponding to the video data, receiving a voice signal, converting the received voice signal into a text, and if a specific text including at least one portion of the converted text exists in the recognized text, sending a message corresponding to a user input of selecting the specific text from the screen corresponding to the video data to the source device.

Abstract: A CPU in a home theater device transmits current sound volume control status data to a digital television receiving device when a theater mode is set. A CPU in the digital television receiving device updates, when the sound volume control status data has been received from the home theater device, sound volume control status data stored in a memory with the received sound volume control status data. A CPU in the digital television receiving device determines sound volume setting value data after an operation based on sound volume operation information fed from a remote control device and the sound volume control status data stored in the memory, and transmits a CEC command including the determined sound volume setting value data to the home theater device while displaying a ratio of the current sound volume to a maximum value in a sound volume adjustable range of the home theater device on a display.

Abstract: Examples include a determination how to manage storage of a video clip generated from recorded video based upon a sensor event. Managing storage of the video clip may include determining whether to save or delete the video clip based on an imprint associated with an object that indicates whether the object is included in the video clip.

Abstract: Disclosed are a method of controlling a source device in a sink device and apparatus therefor. The present invention includes the steps of outputting the video data received from the source device to a display unit, recognizing a text from a screen corresponding to the video data outputted to the display unit, saving the recognized text to a memory to correspond to location information on the screen corresponding to the video data, receiving a voice signal, converting the received voice signal into a text, and if a specific text including at least one portion of the converted text exists in the recognized text, sending a message corresponding to a user input of selecting the specific text from the screen corresponding to the video data to the source device.

Abstract: An apparatus for reconstructing a frame including a speech signal as a reconstructed frame is provided, the apparatus including a determination unit and a frame reconstructor being configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially includes the first reconstructed pitch cycle, such that the reconstructed frame completely or partially includes a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle.

Abstract: A vehicular display device according to the present invention includes a display panel, an information acquiring unit that acquires information on a vehicle; a navigation information acquiring unit that acquires navigation information; and a display controlling unit that displays the information on the vehicle and the navigation information on the display panel. When displaying map information contained in the navigation information behind the information on the vehicle displayed on the display panel, the display controlling unit displays at least two icons indicating positions included in the map information under route guidance in such a manner that the at least two icons do not overlap the information on the vehicle.

Abstract: A method, system, and/or computer program product processes a data plot comprising a plurality of data points for inclusion of additional information content. A space of the data plot is divided into subspaces, where each subspace contains at least one data point of the data plot. An available area for each subspace is computed, and then a compressed information representation for each subspace is computed based on information about said at least one data point contained in said each subspace and a computed available area for said each subspace.

Abstract: Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information.

Abstract: Intensifying contextually relevant sound events for time-delayed broadcast uses multiple directional microphones that capture sound events from a specific location within regions of a stadium. The events are contextually relevant to the environment, such as a referee's whistle sound. A processor executes signal processing of the captured events on each channel to produce audio samples each having a signal profile. The profiles are automatically compared to reference feature templates stored in a database, correspond to pre-identified sound events of contextual relevance. The signal processing accentuates characteristic traits in the events, reflective of contextually relevant events that should be included in a final composite audio output for transmission. If the comparison of samples to the feature templates suggests a high probability of a correspondence, then buffered audio for that channel is introduced into a final audio mix.

Abstract: The invention provides a computer system for interacting with a user. A set of concepts initially forms a target set of concepts. An input module receives a language input from the user. An analysis system executes a plurality of narrowing cycles until a concept packet having at least one concept has been identified. Each narrowing cycle includes identifying at least one portion of the language and determining a subset of concepts from the target set of concepts to form a new target subset. An action item identifier identifies an action item from the action items based on the concept packet. An action executer that executes an action based on the action item that has been identified.

Abstract: A server is connected via a wireless communication network to be capable of receiving sensor information from a wireless device and provide control signals based on at least the sensor information to the wireless device for enabling the wireless device to perform a control task.

Abstract: Aspects of the subject disclosure may include, for example, obtaining data that is captured by a device, obtaining metadata corresponding to the data where the metadata describes a context of the capturing of the data, analyzing the metadata to identify a segment context of a segment of the data without performing an image recognition for the segment of the data, and, responsive to a determination that the context does not satisfy a context criterion, generating a low bandwidth version of the segment and transmitting the low bandwidth version of the segment of the data, wherein the low bandwidth version is transmitted at a second bandwidth that is lower than the first bandwidth. Other embodiments are disclosed.

Abstract: A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Abstract: The processing of voice inputs includes receiving a voice input from a user. The received voice input can be analyzed to determine whether the voice input includes at least one of human-intended content or machine-intended content. Responsive to determining that the voice input includes human-intended content, a human recipient for the human-intended content can be identified within the voice input, and a message can be sent to the identified human recipient. The message can include the human-intended content in an audio form. Responsive to determining that the voice input includes machine-intended content, a machine recipient for the machine-intended content can be identified within the voice input, and a message including the machine-intended content can be sent to the identified machine recipient to implement the machine-intended content.

Abstract: In embodiments, apparatuses, methods and storage media are described that are associated with adjusting audio during content presentation. While content is being presented, one or more persons may be identified that are consuming the content. The persons may be identified via various techniques, such as voice recognition, facial recognition, and distance detection. When persons may be uniquely identified, user audio preferences may be retrieved and used to adjust audio. Audio may be adjusted when persons are not uniquely identified, such as based on a number of persons or their location relative to a content consumption device. Audio adjustment may include volume adjustment and/or adjustment of audio, such as through application of audio effects, as it being presented. Other embodiments may be described and claimed.

Abstract: A base station for use in a communication system is configured to determine whether a set of access information provided by an end unit is acceptable to the base station. The access information facilitates identification by the base station of one or more codecs used by the end unit to encode and decode digital audio signals. Other end units may use different codecs to encode and decode digital audio signals. The base station is configured to simultaneously handle multiple different codecs. The communication system also comprises at least one end unit configured to determine the codec used by the base station, request registration to the base station, and set the determined codec as the codec to use for communications with the base station.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that results from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets (412, 413) is described. A lost packet (412) is a packet which is deemed to be lost by a transform-based audio decoder. Each of the one or more lost packets (412, 413) comprises a set of transform coefficients (313). A set of transform coefficients (313) is used by the transform-based audio decoder to generate a corresponding frame (412, 413) of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet (412) of the one or more lost packets (412, 413) a number of preceding lost packets from the one or more lost packets (313); wherein the determined number is referred to as a loss position.

Abstract: Provided is a method and apparatus for determining a signal coding mode. The signal coding mode may be determined or changed according to whether a current frame corresponds to a silence period and by using a history of speech or music presence possibilities.

Abstract: An apparatus includes a user input unit, a display unit, a control unit, and a buffer unit. The display unit includes a speed setting menu. The control unit selects a mode from the speed setting menu in response to the selection signal of the user, and controls a compression ratio of a voice codec and a transfer rate of a modem corresponding to a transmission-side radio, and a reception rate of a modem and a restoration rate of a voice codec corresponding to a reception-side radio, based on the selected mode. The buffer unit performs a storage function if there is a difference between the compression ratio of the voice codec and the transfer rate of the modem or if there is a difference between the reception rate of the modem and the restoration rate of the voice codec.

Abstract: Systems and processes for natural language processing are provided. In accordance with one example, a method includes, at a first electronic device with one or more processors and memory, receiving a plurality of words, mapping each of the plurality of words to a word representation, and associating the mapped words to provide a plurality of phrases. In some examples, each of the plurality of phrases has a representation of a first type. The method further includes encoding each of the plurality of phrases to provide a respective plurality of encoded phrases. In some examples, each of the plurality of encoded phrases has a representation of a second type different than the first type. The method further includes determining a value of each of the plurality of encoded phrases and identifying one or more phrases of the plurality of encoded phrases having a value exceeding a threshold.

Abstract: Methods and devices for encoding a sparse signal x to generate a compressed encoded signal. The methods employ directionless grammar-based dimension coding. Using labelled subsets and the finding of disjoint repeated subsets in order to build a directionless grammar, the non-zero positions of the sparse signal are encoded in a directionless grammar-based dimension encoder. Element values are encoded in a conditional non-zero encoder. The coding process facilitates random access.

Abstract: The invention relates to method applicable in a packet-switched cellular network for voice connections, by which method a new dedicated channel can be established quickly using the existing dedicated channel of the voice connection. The establishment of the new dedicated channel utilizes the previous dedicated channel the existence of which is extended by sending to at least the receiving terminals post-speech packets according to the invention. The post-speech packets produce a 2-second delay, approximately, which is sufficient to establish a new dedicated channel between terminals participating in a session.