摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
A communication system includes an endpoint that performs codec selection based on at least one network parameter. In a particular embodiment, a communication session exchanges voice information, and the codec selection improves the overall voice quality of the communication session.
摘要:
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.
摘要:
A system and method for identifying a participant during a conference call include the capability to receive a packet containing data that represents audible sounds spoken by one of a plurality of participants in a conference call and to determine a speaker of the audible sounds using voice profile information of the participants. The system and method further include the capability to provide identification information of the speaker to the other participants in the conference call contemporaneously with providing audible sounds based on the data to those participants.
摘要:
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—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.
摘要:
A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.