APPLICATION AWARE RATE CONTROL

• US 20090164657A1
• Filed: 12/20/2007
• Published: 06/25/2009
• Est. Priority Date: 12/20/2007
• Status: Abandoned Application

• Current Assignee: Microsoft Technology Licensing LLC (Microsoft Corporation)
• Original Assignee: Microsoft Corporation
• Inventors: Chen, Minghua, Li, Jin, Chou, Philip A.
• Application Number:
  US11/961,900
• Publication Number:
  US 20090164657A1
• Time in Patent Office: Days
• US Class Current: 709/233
• CPC Class Codes: H04L 41/0896 Bandwidth or capacity management, i.e. automatically increasing or decreasing capacities flow or congestion control using dynamic resource allocation, e.g. in-call renegotiation, H04L47/76
• CPC Class Codes: H04L 43/0864 Round trip delays
• CPC Class Codes: H04L 43/087 Jitter
• CPC Class Codes: H04L 43/0882 Utilisation of link capacity
• CPC Class Codes: H04L 43/10 Active monitoring, e.g. heartbeat, ping or trace-route
• CPC Class Codes: H04L 43/16 Threshold monitoring
• CPC Class Codes: H04L 65/752 adapting media to network capabilities
• CPC Class Codes: H04L 65/80 Responding to QoS

Abstract

A “communications rate controller” provides various techniques for maximizing a quality of real-time communications (RTC) (including audio and/or video broadcasts and conferencing) over multi-hop networks such as, for example, the Internet. Endpoints in such networks generally communicate via a segmented path that extends through one or more routers between each endpoint. Maximization of conferencing quality is generally accomplished by providing in-session bandwidth estimation across segments of the network path between endpoints (i.e., communication/conference participants) in combination with a robust non-oscillating dynamic rate control strategy for maximizing usage of available bandwidth between RTC endpoints. Further, the dynamic rate control techniques provided by the communications rate controller are designed to prevent degradation in end-to-end delay, jitter, and packet loss characteristics of the RTC.

Claims

1. A method for performing real-time estimation of available bandwidth between endpoints in a network for dynamically controlling communication data rates, comprising using a computing device for:

establishing a communications session between a first network endpoint and a second network endpoint across a network path including one or more network nodes between the first and second network endpoints;

wherein the communications session includes an ongoing transmission of encoded communications data packets from the first network endpoint to the second network endpoint at a current sending rate;

periodically collecting network statistical information during the communications session;

periodically computing a current packet queuing delay for at least some of the communications data packets transmitted from the first network endpoint to the second network endpoint;

periodically performing a real-time estimate of a current available bandwidth from current network statistical information; and

periodically adjusting the current sending rate to be as close as possible to the current available bandwidth, with the current available bandwidth representing an upper maximum limit on the current sending rate, based on a computed relationship between the current packet queuing delay and an allowable delay threshold.

2. The method of claim 1 wherein the current sending rate is initially determined by automatically increasing the current sending rate, beginning with a minimum current sending rate, until the current packet queuing delay exceeds the allowable delay threshold at any of the network nodes.

3. The method of claim 1 wherein the current sending rate is automatically decreased as soon as the current packet queuing delay exceeds the allowable delay threshold at any of the network nodes.

4. The method of claim 1 wherein the current sending rate is automatically increased whenever the current packet queuing delay is less than the allowable delay threshold for a predetermined period of time.

5. The method of claim 1 wherein the encoded communications data packets includes an encoded audio stream and an encoded video stream or a parity stream.

6. The method of claim 5 wherein the sending rate is divided between the encoded audio stream and the encoded video stream or the parity stream, and wherein a first portion of the sending rate, used for transmission of the encoded audio stream from the first network endpoint to the second network endpoint, is maintained at a constant rate when decreasing the sending rate.

7. The method of claim 1 wherein the encoded communication data packets are encoded using scalable coding having a base layer and one or more enhancement layers, and wherein one or more of the enhancement layers are added to the communications data packets whenever the sending rate is increased.

8. The method of claim 1 wherein the allowable delay threshold is set to ensure acceptable packet loss and jitter control characteristics of at least a portion of the communications data packets.

9. The method of claim 1 wherein the communications data packets include a series of periodic probing packets that are used to generate the network statistical information during the communications session.

10. A process for dynamically controlling a sending rate of a communications session between endpoints in a network, comprising steps for:

(a) establishing a communications session along a network communications path from a first network endpoint and a second network endpoint, said path including one or more network nodes;

(b) setting an acceptable quality level for the communications session;

(c) beginning with an initial sending rate, increasing a current sending rate of the communications session until a current packet queuing delay at the current sending rate at any of the network nodes exceeds the allowable delay threshold;

(d) gathering current network statistical information;

(e) computing an available bandwidth based on the current network statistical information, said statistical information comprising at least the current packet queuing delay;

(f) using a computed relationship between the current packet queuing delay and the allowable delay threshold for setting a real-time communications rate for sending communications data packets from the first network endpoint to the second network endpoint, and using the computed available bandwidth as an upper limit on the real-time communications rate; and

(g) periodically repeating steps (d) through (f) during the communications session to dynamically adjust the real-time communications rate for maximally utilizing available bandwidth between the first network endpoint and the second network endpoint.

11. The process of claim 10 further comprising steps for decreasing the real-time communications rate as soon as the current packet queuing delay exceeds the allowable delay threshold at any of the network nodes.

12. The process of claim 10 further comprising increasing the real-time communications rate whenever the current packet queuing delay is less than the allowable delay threshold at all of the network nodes for a predetermined period of time.

13. The process of claim 10 further comprising steps for setting the allowable delay threshold to ensure acceptable packet loss and jitter control characteristics of at least a portion of the communications data packets.

14. The process of claim 10 wherein the encoded communications data packets includes an encoded audio stream and an encoded video stream or a parity stream.

15. The process of claim 14 wherein the real-time communications rate is divided between the encoded audio stream and the encoded video stream or the parity stream, and wherein a first portion of the real-time communications rate, used for transmission of the encoded audio stream from the first network endpoint to the second network endpoint, is maintained at a constant rate when decreasing the real-time communications rate.

16. A computer-readable medium having computer executable instructions stored thereon for performing in-session bandwidth estimation and rate control during a communications session between network endpoints, comprising instructions for:

setting an allowable delay threshold in a network path between a first network endpoint and a second network endpoint, said path including one or more network nodes;

beginning with an initial current sending rate, increasing the current sending rate of communications data packets from the first network endpoint to the second network endpoint until a current packet queuing delay at the current sending rate at any of the network nodes exceeds the allowable delay threshold;

periodically recomputing the current packet queuing delay;

periodically computing a current available bandwidth using the current sending rate and the current packet queuing delay in combination with periodically collected network statistical information; and

periodically evaluating the current packet queuing delay and adjusting the current sending rate relative to the current available bandwidth.

17. The computer-readable medium of claim 16 further comprising instructions for decreasing the current sending rate as soon as the current packet queuing delay exceeds the allowable delay threshold at any of the network nodes.

18. The computer-readable medium of claim 16 further comprising instructions for increasing the current sending rate whenever the current packet queuing delay is less than the allowable delay threshold at all of the network nodes for a predetermined period of time.

19. The computer-readable medium of claim 16 further comprising instructions for setting the allowable delay threshold to ensure acceptable packet loss and jitter control characteristics of at least a portion of the communications data packets.

20. The computer-readable medium of claim 16 wherein the communications data packets include an encoded audio stream and an encoded video stream or a parity stream, and further comprising instructions for:

dividing the current sending rate between the encoded audio stream and the encoded video stream or the parity stream; and

wherein a first portion of the real-time communications rate, used for transmission of the encoded audio stream from the first network endpoint to the second network endpoint, is maintained at a constant rate when decreasing the current sending rate.