VoIP bandwidth management

• US 9,356,869 B2
• Filed: 04/10/2013
• Issued: 05/31/2016
• Est. Priority Date: 04/10/2013
• Status: Active Grant

• Current Assignee: Viber Media Inc. (Rakuten Group, Inc.)
• Original Assignee: Viber Media Inc. (Rakuten Group, Inc.)
• Inventors: Marueli, Sunny, Shalgi, Ran
• Primary Examiner(s): Lai, Andrew
• Assistant Examiner(s): Andrews, Hoyet H
• Application Number:
  US13/859,765
• Publication Number:
  US 20140307543A1
• Time in Patent Office: 1,147 Days
• US Class Current: 1/1
• CPC Class Codes: H04L 47/11 Identifying congestion
• CPC Class Codes: H04L 47/12 Avoiding congestion; Recovering from congestion
• CPC Class Codes: H04M 3/2236 Quality of speech transmission monitoring
• CPC Class Codes: H04M 7/0084 Network monitoring; Error detection; Error recovery; Network testing
• CPC Class Codes: H04M 7/1275 Methods and means to improve the telephone service quality, e.g. reservation, prioritisation or admission control

Abstract

A computerized method of optimizing audio quality in a voice stream between a sender and a receiver VoIP applications, comprising: defining by the receiver time intervals; determining by the receiver at the end of each time interval whether congestion exists, by calculating (i) one-way-delay and (ii) trend, using double-exponential smoothing; estimating by the receiver a bandwidth available to the sender based on said calculation; sending said estimated bandwidth by the receiver to the sender; and using by the sender said bandwidth estimate as maximum allowed send rate.

Claims

1. A computerized method of optimizing audio quality in a voice stream of Voice over Internet Protocol (VoIP) applications between a sender and a receiver, comprising:

defining by the receiver a plurality of time intervals for the voice stream;

determining by the receiver at an end of each of the plurality of the defined time intervals whether congestion exists, by

(i) calculating a one-way-delay for each of a first plurality of received audio packets in the voice stream and

(ii) using double exponential smoothing to calculate a one-way-delay trend of the first plurality of received audio packets;

estimating by the receiver a bandwidth available to the sender based on a result of said one-way-delay trend calculation;

sending by the receiver said estimated bandwidth to the sender; and

using by the sender said estimated bandwidth received by the sender as maximum allowed send rate for a second plurality of audio packets in the voice stream of the VoIP applications.

2. The method of claim 1, wherein said determining whether congestion exists comprises determining that congestion exists if the calculated one-way-delay is greater than a first predefined positive constant or the calculated one-way-delay trend is greater than a second predefined positive constant.

3. The method of claim 2, further comprising determining congestion level based on a value of said calculated one-way-delay trend.

4. The method of claim 1, additionally comprising a step of determining whether a bandwidth estimation should be done;

and wherein said estimating, sending and using are performed only if it is determined that said bandwidth estimation should be done.

5. The method of claim 4, wherein said determining whether said bandwidth estimation should be done comprises determining whether a predefined time length has elapsed since a previous bandwidth estimation was done.

6. The method of claim 5, wherein said predefined time length is a round trip time.

7. The method of claim 4, wherein said determining whether said bandwidth estimation should be done comprises determining whether congestion state has changed.

8. The method of claim 1, wherein said bandwidth estimation comprises:

a. estimating an incoming bitrate of the voice stream;

b. if it is determined there is no congestion, setting the estimation of the bandwidth at a value greater than the estimated incoming bitrate; and

c. if it is determined there is congestion, setting the estimation of the bandwidth at a value smaller than the estimated incoming bitrate.