Abstract: Methods and systems are disclosed wherein TCP may approximate Reliable Transport Protocol (RTP) or UDP delivery for real-time video/data conferencing applications that have long RTT connections.

Abstract: Presented herein are downstream recovery (error correction) techniques for an aggregated/consolidated media stream. In one example, a consolidated media stream that includes source media packets from one or more sources is sent to one or more downstream receiving devices. Based on the source media packets, one or more self-describing recovery packets for downstream error correction of the source media packets are generated. The self-describing recovery packets include a mapping to the source media packets used to generate the self-describing recovery packets, thereby avoiding the addition of error correction information in the consolidated media stream. The one or more self-describing recovery packets are sent to each of the downstream receiving devices as a separate stream.

Abstract: Embodiments include detecting an increase in delay of a flow assigned to a first queue of a network device, where the increase is sufficient to cause the flow rate of the flow to decrease if the flow is delay-sensitive. Embodiments further include determining whether an amount of bandwidth consumed by the flow decreases sufficiently after the increase is detected, and assigning the flow to a second queue based, at least in part, on determining the amount of bandwidth consumed by the flow does not decrease sufficiently. Specific embodiments include evaluating a series of two or more bandwidth measurements of the flow according to a bandwidth reduction measure to determine whether the amount of bandwidth consumed by the flow decreases by sufficiently. More specific embodiments include the first queue being configured to receive delay-sensitive flows and the second queue being configured to receive delay-insensitive nice flows.

Abstract: A communication device detects sound with a microphone to produce a sound signal representative of the sound and searches the sound signal for unique inaudible sound signatures associated with a group identifier. Each sound signature identifies a respective communication device. Each sound signature is in an audible frequency band but masked to be imperceptible. If any of the sound signatures are determined to be present in the sound signal, the communication device selects one based on predetermined criteria associated with the group identifier.

Abstract: In one method embodiment, providing a multiplex of compressed versions of a first video stream and a first audio stream, each corresponding to an audiovisual (A/V) program, the first video stream and the first audio stream each corresponding to a first playout rate and un-synchronized with each other for an initial playout portion; and providing a compressed version of a second audio stream, the second audio stream corresponding to a pitch-preserving, second playout rate different than the first playout rate, the second audio stream synchronized to the initial playout portion of the first video stream when the first video stream is played out at the second playout rate, the first audio stream replaceable by the second audio stream for the initial playout portion.

Abstract: Presented herein are techniques for receiving media at a participant device for a conference session. The media may be transmitted to the participant device using a first transport mechanism. A command is received from a user of a participant device in a conference session to playback a portion of the conference session. A request is sent to the conference server to retransmit a recording of the portion of the conference session requested for playback. Data is received for the recording of the portion of the conference session transmitted from the conference server using a second transport mechanism, wherein the second transport mechanism is a higher reliability transport mechanism than the first transport mechanism.

Abstract: A system includes a processor operable to identify each of a plurality of first video packets from a communication session with a first identifier designating that bandwidth is reserved on the network for the packet, and identify each of a plurality of second video packets from the communication session with a second identifier designating that bandwidth is not reserved on the network for the packet. The system further includes an interface operable to initiate transmission of the plurality of first video packets and the plurality of second video packets in the communication session.

Abstract: Presented herein are downstream recovery (error correction) techniques for an aggregated/consolidated media stream. In one example, a consolidated media stream that includes source media packets from one or more sources is sent to one or more downstream receiving devices. Based on the source media packets, one or more self-describing recovery packets for downstream error correction of the source media packets are generated. The self-describing recovery packets include a mapping to the source media packets used to generate the self-describing recovery packets, thereby avoiding the addition of error correction information in the consolidated media stream. The one or more self-describing recovery packets are sent to each of the downstream receiving devices as a separate stream.

Abstract: Presented herein are techniques for receiving media at a participant device for a conference session. The media may be transmitted to the participant device using a first transport mechanism. A command is received from a user of a participant device in a conference session to playback a portion of the conference session. A request is sent to the conference server to retransmit a recording of the portion of the conference session requested for playback. Data is received for the recording of the portion of the conference session transmitted from the conference server using a second transport mechanism, wherein the second transport mechanism is a higher reliability transport mechanism than the first transport mechanism.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: In one embodiment, a system and method include determining bandwidth of a link that connects a local modem to a remote router. A first percentage of the bandwidth is assigned to a first class of data and a second percentage of bandwidth is assigned to a second class of data. The remaining percentage of the bandwidth is assigned for nominal excess capacity. The flow of first class of data and second class of data are controlled to below respective percentages of the bandwidth.

Abstract: A method that includes; (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: A method is provided in one example embodiment that includes monitoring a sound pressure level with an endpoint (e.g., an Internet Protocol (IP) phone), which is configured for communications involving end users; analyzing the sound pressure level to detect a sound anomaly; and communicating the sound anomaly to a sound classification module. The endpoint can be configured to operate in a low-power mode during the monitoring of the sound pressure level. In certain instances, the sound classification module is hosted by the endpoint. In other implementations, the sound classification module is hosted in a cloud network.

Abstract: According to certain embodiments, training a transcription system includes accessing recorded voice data of a user from one or more sources. The recorded voice data comprises voice samples. A transcript of the recorded voice data is accessed. The transcript comprises text representing one or more words of each voice sample. The transcript and the recorded voice data are provided to a transcription system to generate a voice profile for the user. The voice profile comprises information used to convert a voice sample to corresponding text.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: A method that includes: (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: In one embodiment, a system and method include determining bandwidth of a link that connects a local modem to a remote router. A first percentage of the bandwidth is assigned to a first class of data and a second percentage of bandwidth is assigned to a second class of data. The remaining percentage of the bandwidth is assigned for nominal excess capacity. The flow of first class of data and second class of data are controlled to below respective percentages of the bandwidth.

Abstract: In an example embodiment, a control connection is introduced between an adaptive jitter buffer (AJB) and an adaptive bulk delay (ABD) buffer of an echo canceller (ECAN) in an Internet Protocol (IP) conference bridge. The control connection allows the AJB to control the amount of delay inserted by the ABD in the ECAN convolution processor (CP) signal path. The adjustment in ABD delay restores the time alignment of the ECAN internal echo estimate and offsets variations in echo tail delay introduced by network induced AJB delay changes. Time-invariance is preserved in the echo tail path.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: A system includes a processor operable to identify each of a plurality of first video packets from a communication session with a first identifier designating that bandwidth is reserved on the network for the packet, and identify each of a plurality of second video packets from the communication session with a second identifier designating that bandwidth is not reserved on the network for the packet. The system further includes an interface operable to initiate transmission of the plurality of first video packets and the plurality of second video packets in the communication session.