User controlled audio quality for voice-over-IP telephony systems
First Claim
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1. A method for controlling a Voice over Internet Protocol (VoIP) call at a telephone endpoint, comprising:
- determining a current payload size for audio packets in the VoIP call;
monitoring a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call;
determining a current jitter amount for the VoIP call; and
changing the current payload size for the audio packets mid-call to correspond with the desired level of user perceived audio quality only when the current jitter amount is less than a predetermined jitter amount that is preset and represents an acceptable perceivable sound quality.
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Abstract
A call adaptation system tracks adaptation schemes used for transmitting audio packets in a Voice Over IP call. A user response to the Voice Over IP (VoIP) call is monitored. The call adaptation system then dynamically varies the adaptation schemes used for transmitting the audio packets according to the monitored user response.
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Citations
52 Claims
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1. A method for controlling a Voice over Internet Protocol (VoIP) call at a telephone endpoint, comprising:
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determining a current payload size for audio packets in the VoIP call; monitoring a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call; determining a current jitter amount for the VoIP call; and changing the current payload size for the audio packets mid-call to correspond with the desired level of user perceived audio quality only when the current jitter amount is less than a predetermined jitter amount that is preset and represents an acceptable perceivable sound quality. - View Dependent Claims (2, 3, 4, 5, 6, 7, 8, 9)
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10. An apparatus, comprising:
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one or more processors; and a memory coupled to the processors comprising instructions executable by the processors, the processors operable when executing the instructions to; determine a current payload size for audio packets in the Voice over Internet Protocol (VoIP) call; monitor a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call; determine a current jitter amount for the VoIP call; and change the current payload size for the audio packets mid-call to correspond with the desired level of user perceived audio quality only when the current jitter amount is less than a predetermined jitter amount that is preset and represents an acceptable perceivable sound quality. - View Dependent Claims (11, 12)
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13. An adaptation system, comprising:
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an input to detect a user response requesting a different user perceived sound quality for a call and specifying a user-defined codec selection; and a controller configured to measure current network congestion of a network that transmits the call and determine a packet loss ratio associated with the requested different user perceived sound quality, the user-defined codec selection and the measured current network congestion; the controller to compare the determined packet loss ratio to a predetermined packet loss ratio that is set according to an empirical analysis identifying a threshold amount of packet loss that represents a minimum call sound quality; the controller to dynamically vary a codec used to encode the call while the call is in progress according to the user-defined codec selection when the determined packet loss ratio is less than the predetermined packet loss ratio; the controller to determine a system-defined codec selection corresponding to both the measured network congestion and the determined packet loss ratio when the determined packet loss ratio is equal to or greater than the predetermined packet loss ratio; and the controller to dynamically vary the codec used to encode the call while the call is in progress according to the system-defined codec selection when the determined packet loss ratio is equal to or greater than the predetermined packet loss ratio. - View Dependent Claims (14, 15, 16, 17, 18, 19, 20, 21, 22)
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23. An electronic storage medium containing software used for controlling a Voice over Internet Protocol (VoIP) call, the software in the electronic storage medium comprising:
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code for determining a current payload size for audio packets in the VoIP call; code for monitoring a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call; code for determining a current jitter amount for the VoIP call; code for changing the current payload size for the audio packets mid-call to correspond with the desired level of user perceived audio quality only when the current jitter amount is less than a predetermined jitter amount that is preset and represents an acceptable perceivable sound quality. - View Dependent Claims (24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34)
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35. A system for controlling a Voice over Internet Protocol call, comprising:
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means for tracking adaptation schemes used for transmitting audio packets in a VoIP call; means for monitoring a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call; means for dynamically varying the adaptation schemes used for transmitting the audio packets from a telephone endpoint in response to user indications so that the user perceived audio quality of the VoIP call corresponds with the monitored user response; means for measuring a first packet loss rate for the VoIP call before dynamically varying the adaptation schemes; means for measuring a second packet loss rate for the VoIP call after dynamically varying the adaptation schemes; means for comparing the first packet loss rate to the second packet loss rate; and means for automatically dynamically re-varying the adaptation schemes to lower bandwidth consumption in response to a determination that the second packet loss rate is a predetermined amount higher than the first packet loss rate. - View Dependent Claims (36, 37, 38, 39, 40, 41, 42)
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43. A system for controlling a Voice over Internet Protocol (VoIP) call, comprising:
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means for tracking packet payload size for a VoIP call; means for monitoring a user response to the VoIP call indicating delays associated with the VoIP call; means for determining current network congestion; means for dynamically varying the packet payload size during the VoIP call so that packets of the VoIP call include payloads of varying sizes according to the monitored user response when the current network congestion is lower than a predetermined threshold; means for initially transmitting the packets using a best effort transmission scheme; means for monitoring the user response for a request to increase voice quality; and means for requesting reservation of network resources for the call during the already established VoIP call when the increase voice quality request is detected from the user response and when the current network congestion is below a predetermined threshold; and means for indicating that the encoding process has not been dynamically varied when the current network congestion is not lower than a predetermined threshold. - View Dependent Claims (44, 45, 46)
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47. An apparatus comprising:
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one or more processors; and a memory coupled to the processors comprising instructions executable by the processors, the processors operable when executing the instructions to; track adaptation schemes used for transmitting audio packets in a Voice over Internet Protocol (VoIP) call; monitor a user response to the VoIP call indicating a desired level of user perceived audio quality for the VoIP call; dynamically vary the adaptation schemes used for transmitting the audio packets from a telephone endpoint in response to user indications so that the user perceived audio quality of the VoIP call corresponds with the monitored user response; measure a first packet loss rate for the VoIP call before dynamically varying the adaptation schemes; measure a second packet loss rate for the VoIP call after dynamically varying the adaptation schemes; compare the first packet loss rate to the second packet loss rate; and automatically dynamically re-vary the adaptation schemes to lower bandwidth consumption in response to a determination that the second packet loss rate is a predetermined amount higher than the first packet loss rate. - View Dependent Claims (48, 49, 50, 51)
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52. A method for controlling a Voice Over Internet Protocol (VoIP) call, comprising:
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tracking adaptation schemes used for transmitting packets in the VoIP call; monitoring a user response to the VoIP call that requests a different level of user perceived sound quality for the VoIP call; and dynamically varying the adaptation schemes used for transmitting the packets in the VoIP call to correspond with the requested level of user perceived sound quality; and wherein dynamically varying the adaptation schemes includes adjusting Forward Error Correction (FEC) and adjusting packet payload length; wherein the packet payload length is dynamically varied according to measured network congestion; wherein the packet payload length is adjusted in response to a user indication of delays during the VoIP call; wherein the packet payload length is increased in response to the user indication, wherein the packet payload length is associated with a first packet payload type having 20 bytes when the delay is less than a predetermined threshold and the packet payload length is associated with a second packet payload type having at least 40 bytes when the delay is greater than the predetermined threshold.
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Specification