Cisco Technology, Inc.

Cisco Systems Inc.

Michael E Knappe

Shmuel Shaffer

Luke K Surazski

Pascal Huart

Joseph Khouri

David R Oran

Robert T Bell

Paul S Hahn

Richard A Dunlap

Cary W FitzGerald

Kevin J Connor

Ramanathan T Jagadeesan

John F Wakerly

Robert E Gleichauf

Lukasz K Surazski

According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Method and system for managing time-sensitive packetized data streams at a receiver

A method and system for logging voice quality issues for a communication connection includes receiving a signal for logging quality information for a voice connection at an endpoint of the voice connection. Voice samples are collected from the voice connection at the endpoint. The voice samples are stored in an error log at the endpoint.

Method and system for logging voice quality issues for communication connections

A system and method are disclosed for interleaving time-critical packets and lower-priority packets onto a common data link. A packet arrival prediction mechanism predicts when a time-critical packet is expected to arrive. When transmission of a waiting lower-priority packet might cause a substantial delay in the expected time-critical packet's transmission, the lower-priority packet is parked until it can be transmitted without interfering with a time-critical packet.

Time-sensitive-packet jitter and latency minimization on a shared data link

Devices, softwares and methods generate, in real time, indexing metadata for select portions of a telephone conversation or conference. The indexing metadata is generated responsive to inputs received while the conversation is being recorded live. The inputs are either by a user pressing a soft key on a telephone device, or by a voice conference bridge determining who is the dominant speaker in a multi-party conference.

Devices, methods and software for generating indexing metatags in real time for a stream of digitally stored voice data

Methods, devices, and software are provided for generating and sending data packets that contain redundant voice data over VoIP connections made under an unreliable network protocol. The redundant data is packaged either in redundant data packets, or in expanded original packets, to repeat data that originally belongs in other packets. Generation of the redundant voice data is either from the transmitting device or from a retransmitting device, such as a router in the network. Generation is triggered either when errors are detected, or simply when the network resources permit it, or both. The received voice data is processed by the second party to the connection, which is typically a telephone call. The redundant voice data that is actually received is discarded. The invention thus ensures that less voice data is lost than in the prior art over VoIP connections made under an unreliable network protocol.

Methods, devices and software for redundant transmission of voice data over a packet network connection established according to an unreliable communication protocol

According to one embodiment of the invention, a method for managing communication impairments between Internet Protocol devices is provided. The method includes determining a transmission of a signal comprising a comfort noise, where the signal is transmitted from a first endpoint to a second endpoint. The method also includes sending a notice signal from the first endpoint to the second endpoint indicating that the signal is transmitted. The method also includes suppressing the signal at the second endpoint in response to the notice signal. According to another embodiment of the invention, a method for managing communication impairments between an Internet Protocol ('IP') phone and an IP device is provided. The method provides sending a status signal to the device indicating that the phone is operating as a speakerphone. The method also includes suppressing the transmission of any comfort noise to the phone in response to the status signal.

Method and system for managing erroneous attenuation of signal

A method and system for participant control of privacy during a multiparty communication session includes receiving a request from a first participant to a multiparty communication connection for a sidebar between the first participant and a second participant to the multiparty communication connection. The sidebar is provided by at least substantially eliminating voice streams generated by the first participant and the second participant from conference output streams generated for a set of remaining participants to the multiparty communication connection.

Method and system for participant control of privacy during multiparty communication sessions

Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter&#x2014;once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.

System and method for stereo conferencing over low-bandwidth links

A system and method for improving the intelligibility of a moderator during a multi-party communication session includes receiving a plurality of participant voice streams from a plurality of respective conference participants. An incoming moderator voice stream may be received from a moderator. The plurality of participant voice streams and the moderator voice stream are transmitted such that the intelligibility of the moderator voice stream is improved relative to at least one of the participant voice streams.

Method and system for improving the intelligibility of a moderator during a multiparty communication session

A method for synchronizing media in a call includes receiving a first input stream of packets of a first media type at a first call resource, generating a first output stream in response to the media streams, receiving synchronization information from a second call resource, and communicating the output stream from the first call resource to a endpoint in synchronization with the second call resource. A video conference bridge includes a first interface operable to receive a video input stream, a processor operable to generate video output, and a second interface operable to receive synchronization information from an audio conference bridge.

Method and apparatus for adding functionality to an existing conference call

A network device and its mode of operation. The network device includes an input port through which audio streams may be received. A controller receives a user input indicating which type of mix the user wants applied to the audio stream and applies the mixing factor designated. The mixing factor may be a stereo mix, a panned mix, or a mono mix. The network device also includes an output port through which the mixed audio stream can be transmitted to the user.

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Sunnyvale, CA, United States of America

Michael E Knappe