System and method for providing a replacement packet

• US 8,879,464 B2
• Filed: 01/29/2009
• Issued: 11/04/2014
• Est. Priority Date: 01/29/2009
• Status: Active Grant

• Current Assignee: Arlington Technologies, LLC (Dominion Harbor Enterprises, LLC)
• Original Assignee: Avaya Incorporated
• Inventors: Michaelis, Paul Roller, Toennis, Roger, Grover, Douglas M.
• Primary Examiner(s): Mattis, Jason
• Assistant Examiner(s): Clawson, Stephen J
• Application Number:
  US12/361,950
• Publication Number:
  US 20100188967A1
• Time in Patent Office: 2,105 Days
• Field of Search: 370/216
• US Class Current: 370/316
• CPC Class Codes: G06F 11/2007 using redundant communication media
• CPC Class Codes: H04L 1/22 using redundant apparatus to increase reliability
• CPC Class Codes: H04L 2001/0097 Relays

Abstract

The system generates a first data stream which represents a data signal. The first data stream is encoded via a first encoding technique. The first data stream comprises one or more packets with a duration and timestamp. The data signal is encoded into a second data stream using a different encoding technique with a corresponding packet in the second data stream. A packet in the second data stream has the same duration and timestamp as the corresponding packet in the first data stream.

The first data stream is sent on a first network. The second data stream is sent on a second network. The first and second data streams are received. A missing/delayed packet in the first data stream is detected. The data signal is regenerated using the first data stream and the packet from the second data stream that corresponds to the missing/delayed packet in the first data stream.

Claims

1. A system for providing a replacement packet comprising:

a. source node configured to generate a first audio stream representing an audio signal, wherein the first audio stream is encoded via a first encoding technique, wherein the first audio stream comprises one or more packets, each with a duration and a timestamp, to encode the represented audio signal into a second audio stream via a second encoding technique that is different from the first encoding technique and that uses less bandwidth than the first encoding technique, wherein the source node is a common source for both the first and second audio streams, wherein for each packet in the first audio stream there is a corresponding packet in the second audio stream, wherein each packet in the second audio stream has the same duration and the same timestamp as the corresponding packet in the first audio stream, to send the first audio stream on a first network, and to send the second audio stream on a second network, wherein the second network comprises a link established over the second network to transmit the second audio stream that provides a lower bandwidth than a link established over the first network to transmit the first audio stream; and

b. a destination node configured to receive the first and second audio streams, to detect a missing or delayed packet in the first audio stream, and to regenerate the represented audio signal using the one or more packets from the first audio stream and a packet corresponding to the missing or delayed packet in the first audio stream from the second audio stream.

2. The system of claim 1, wherein generating the first audio stream representing the audio signal is accomplished by a technique selected from the group comprising:

encoding the audio signal or forwarding encoded packets of the audio signal.

3. The system of claim 1, wherein the destination node is further configured to detect that the corresponding packet to the missing or delayed packet in the first audio stream was not received and improve a perceived quality of the regenerated audio signal by replacing audio from the missing or delayed packet in the first audio stream using an interpolation process selected from the group comprising:

interpolating between packets, filling with white noise, and filling with pink noise.

4. The system of claim 1, wherein the one or more packets in the first audio stream comprise a plurality of packets, wherein the duration of at least two packets of the plurality of packets in the first audio stream is different, wherein the one or more packets in the second audio stream comprise a plurality of packets, and wherein the duration of at least two packets of the plurality of packets in the second audio stream is different.

5. The system of claim 1, wherein the one or more packets in the first audio stream or the second audio stream contain at least one packet type selected from the group comprising:

a SIP packet, a H.323 packet, and an ISDN packet.

6. The system of claim 1, wherein the one or more corresponding packets in the second audio stream are encoded using a technique that requires an audio rate less than or equal to 32 kilobits per second.

7. The system of claim 1, wherein the first audio stream is encoded with a technique that provides a Mean Opinion Score (MOS) rating equal to or better than the MOS for ITU-T Recommendation G.711.

8. The system of claim 1, wherein the first network is a wired network and the second network is a wireless network.

9. The system of claim 8, wherein the corresponding packets in the second audio stream are packetized using at least one protocol selected from the group comprising:

GSM, CDMA, TDMA, and IEEE 802.11 protocols.

10. The system of claim 1, wherein

the source node is configured to encode at least one threshold point in the represented audio signal into at least one corresponding threshold packet that has a duration and a timestamp, wherein the corresponding packet in the first audio stream has the same duration and the same timestamp as the at least one threshold packet.

11. The system of claim 10, wherein the destination node is further configured to detect a missing or delayed packet in the first audio stream that does not correspond to the at least one threshold packet, and improve a perceived quality of the regenerated audio signal by replacing audio in the missing or delayed packet using an interpolation process selected from the group comprising:

interpolating between packets, filling with white noise, and filling with pink noise.

12. The system of claim 10, wherein the at least one threshold point is encoded based on at least one technique selected from the group comprising:

a psycho-acoustic prioritization in which a priority score is based on the audio signal'"'"'s contribution to intelligibility, a psychoacoustic prioritization in which the priority score is based on the audio signal'"'"'s contribution to perceived a voice quality, presence or absence of a within-sample plosive sound, and a transition from one type of speech sound to another.

13. The system of claim 10, wherein the source node is further configured to set a threshold level for the at least one threshold packet based on at least one item selected from the group comprising:

a change in a quality of service of the first network, a change in a quality of service for the second network, a bandwidth of a link on the second network, a grouping of audio streams, and a change in the a number of audio streams being sent by the source node.

14. The system of claim 10, wherein

the source node is configured to encode a threshold point of one or more threshold points in each of a plurality of represented audio signals into a separate corresponding threshold packet each with a duration and a timestamp, wherein for each threshold packet there is a corresponding packet in the corresponding first and second audio streams, and wherein each threshold packet has the same duration and the same timestamp as the corresponding packet in the corresponding first and second audio streams.

15. The system of claim 14, wherein the threshold packets are multiplexed over a common link within the second network.

16. The system of claim 14, wherein encoding of the threshold packets are based on a quality of the first network or a quality of the second network.

17. The system of claim 14, wherein encoding of the threshold packets is based on a number of the audio streams.

18. The system of claim 14, wherein the thresholds packets are multiplexed over a common link within the second network.

19. The system of claim 14, wherein encoding of the threshold packets is based on a quality of the first network or a quality of the second network.

20. The system of claim 14, wherein encoding of the threshold packets is based on the number of separate audio streams.

21. The system of claim 14, wherein the plurality of separate audio streams are multiplexed into a single audio stream.

22. The system of claim 14, wherein the source node is further configured to set a threshold level for the corresponding threshold packets based on at least one item selected from the group comprising:

a change in a quality of service of the first network, a change in a quality of service for the second network, a bandwidth of a link on the second network, a grouping of audio streams, and a change in the a number of audio streams being sent by the source node.

23. A method for providing a replacement packet comprising:

a. generating, by a common source node, a first audio stream representing an audio signal, wherein the first audio stream is encoded using a first encoding technique, and wherein the first audio stream comprises one or more packets, each with a duration and a timestamp;

b. generating, by the common source node, a second audio stream representing an audio signal, wherein the second audio stream is encoded using a second encoding technique, and wherein the second audio stream comprises one or more packets, each with a duration and a timestamp, the second encoding technique using less bandwidth than the first encoding technique, the second audio stream having a corresponding packet for each packet in the first audio stream that has the same duration and the same timestamp as the corresponding packet in the first audio stream, wherein the second network comprises a link established over the second network that provides a lower bandwidth to transmit the second audio stream when compared to a link established over the first network to transmit the first audio stream;

c. sending, by the common source node, the first audio stream on the first network;

d. sending, by the common source node, the second audio stream on the second network;

e. receiving, by a common destination node, the first and second audio streams;

f. detecting, by the common destination node, a missing or delayed packet in the first audio stream; and

g. regenerating, by the common destination node, the represented audio signal using the one or more packets from the first audio stream and the corresponding packet to the missing or delayed packet from the second audio stream.

24. The method of claim 23, wherein generating the first data stream representing the data signal is accomplished by at least one technique selected from the group comprising:

encoding the data signal or forwarding encoded packets of the data signal.

25. The method of claim 23, wherein the step of regenerating the data signal further comprises the steps of:

h. detecting that the packet in the second data stream corresponding to the missing or delayed packet was not received; and

i. improving a quality of the regenerated data signal by replacing data from the missing or delayed packet in the first data stream using an interpolation process selected from the group comprising;

interpolating between packets, filling with white noise, and filling with pink noise.

26. The method of claim 23, wherein the one or more packets in the first data stream comprise a plurality of packets, and wherein the duration of at least two packets of the plurality of packets is different.

27. The method of claim 23, wherein the one or more packets in the first data stream or the second data stream contain at least one packet type selected from the group comprising:

a SIP packet, a H.323 packet, and an ISDN packet.

28. The method of claim 23, wherein the one or more corresponding packets in the second data stream are encoded using a technique that requires a data rate less than or equal to 32 kilobits per second.

29. The method of claim 23, wherein the first data stream is encoded with a technique that provides a Mean Opinion Score rating equal to or better than the MOS for ITU-T Recommendation G.711.

30. The method of claim 23, wherein the first network is a wired network and the second network is a wireless network.

31. The method of claim 30, wherein the one or more packets in the second data stream are packetized using at least one protocol selected from the group comprising:

GSM, CDMA, TDMA, and IEEE 802.11 protocols.

32. The method of claim 23, wherein the common source node

encodes at least one threshold point in the represented audio signal, each into a corresponding threshold packet that has a duration and a timestamp, wherein for each threshold packet there is a corresponding packet in the first audio stream that has the same duration and the same timestamp as the corresponding threshold packet.

33. The method of claim 32, wherein generating the first audio stream representing the audio signal is accomplished by a technique selected from the group comprising:

encoding the audio signal or forwarding encoded packets of the audio signal.

34. The method of claim 32, further comprising the steps of:

h. detecting a missing or delayed packet in the first audio stream that does not correspond to a threshold packet; and

i. improving a quality of the regenerated audio signal by replacing audio in the missing or delayed packet using an interpolation process selected from the group comprising;

interpolating between packets, filling with white noise, and filling with pink noise.

35. The method of claim 32, wherein the one or more packets in the first audio stream comprise a plurality of packets, and wherein the duration of at least two packets of the plurality of packets is different.

36. The method of claim 32, wherein the at least one threshold point is encoded based on at least one technique selected from the group comprising:

a psycho-acoustic prioritization in which a priority score is based on the audio signal'"'"'s contribution to intelligibility, a psychoacoustic prioritization in which the priority score is based on the audio signal'"'"'s contribution to a perceived voice quality, the presence or absence of a within-sample plosive sound, and a transition from one type of speech sound to another.

37. The method of claim 32, further comprising the step of setting a threshold level for the at least one threshold packet based on at least one item selected from the group comprising:

a change in a quality of service of the first network, a change in a quality of service for the second network, a bandwidth of a link on the second network, a grouping of audio streams, and a change in the a number of audio streams being sent by the source node.

38. The method of claim 23, wherein the common source node

encodes at least one threshold point in each of a plurality of audio signals into a threshold packet for each threshold point, each with a duration and a timestamp, wherein for each threshold packet there is a corresponding packet in corresponding first and second audio streams, and wherein each corresponding packet in each corresponding audio stream has the same duration and the same timestamp as each corresponding threshold packet.

39. The method of claim 38, wherein generating the first audio stream representing the audio signal is accomplished by a technique selected from the group comprising:

encoding the audio signal or forwarding encoded packets of the audio signal.

40. The method of claim 38, wherein each of the corresponding thresholds packets are multiplexed over a common link within the second network.

41. The method of claim 38, wherein encoding each of the threshold packets is based on a quality of the first network or a quality of the second network.

42. The method of claim 38, wherein encoding each of the threshold packets is based on the number of separate audio streams.

43. The method of claim 38, wherein the plurality of separate audio streams are multiplexed into a single audio stream.

44. The method of claim 38, further comprising the step of setting a threshold level for the corresponding threshold packets based on at least one item selected from the group comprising:

a change in a quality of service of the first network, a change in a quality of service for the second network, a bandwidth of a link on the second network, a grouping of audio streams, and a change in the a number of audio streams being sent by the source node.

45. A system for providing a replacement packet comprising:

a. a source node configured to generate a first data stream representing a data signal, wherein the first data stream is encoded via a first encoding technique, wherein the first data stream comprises one or more packets, each with a duration and a timestamp, to encode the represented data signal into a second data stream via a second encoding technique that is different from the first encoding technique and that uses less bandwidth than the first encoding technique, wherein for each packet in the first data stream there is a corresponding packet in the second data stream, wherein each corresponding packet in the second data stream has the same duration and the same timestamp as the corresponding packet in the first data stream, to send the first data stream on a first network, and to send the second data stream on a second network, wherein the second network comprises a link established over the second network that provides a lower bandwidth than a link established over the first network.

46. A system for providing a replacement packet comprising:

a. a destination node configured to receive an encoded first data stream which represents a data signal, wherein the first data stream comprises one or more packets, each with a duration and a timestamp, to receive a second data stream, wherein the second data stream is encoded from the represented data signal using a different encoding technique than the first data stream, wherein the different encoding technique uses less bandwidth than the first encoding technique, wherein for each packet in the first data stream there is a corresponding packet in the second data stream, wherein each packet in the second data stream has the same duration and timestamp as the corresponding packet in the first data stream, and wherein the second data stream was sent over a second different network, the second different network comprises a link established over the second different network that provides a lower bandwidth than a link established over the first network, to detect a missing or delayed packet in the first data stream, and to regenerate the represented data signal using the one or more packets from the first data stream and a packet corresponding to the missing or delayed packet in the first data stream from the second data stream.

47. The system of claim 46, further comprising

a source node configured to generate the first data stream representing the data signal, wherein the first data stream is encoded via the first encoding technique, and -encode at least one threshold point in the represented data signal into at least one corresponding threshold packet that has a duration and a timestamp, wherein the corresponding packet in the first audio stream has the same duration and the same timestamp as the at least one threshold packet.

48. The system of claim 46, wherein the

first data stream represents an audio stream, wherein at least one threshold packet is generated from the represented audio stream, and wherein the at least one threshold packet has a corresponding packet in the first audio stream and has the same duration and timestamp of the corresponding packet in the first audio stream.

49. A tangible and non-transient computer readable medium comprising microprocessor executable instructions that, when executed, perform steps comprising:

generating, by a common source node, a first audio stream representing an audio signal, wherein the first audio stream is encoded using a first encoding technique, and wherein the first audio stream comprises one or more packets, each with a duration and a timestamp;

generating, by the common source node, a second audio stream representing an audio signal, wherein the second audio stream is encoded using a second encoding technique, and wherein the second audio stream comprises one or more packets, each with a duration and a timestamp, the second encoding technique using less bandwidth than the first encoding technique, the second audio stream having a corresponding packet for each packet in the first audio stream that has the same duration and the same timestamp as the corresponding packet in the first audio stream, wherein the second network comprises a link established over the second network that provides a lower bandwidth to transmit the second audio stream when compared to a link established over the first network to transmit the first audio stream;

sending, by the common source node, the first audio stream on the first network; and

sending, by the common source node, the second audio stream on the second network.

50. The computer readable medium of claim 49, wherein a common destination node receives the first and second audio streams;

detects a missing or delayed packet in the first audio stream; and

regenerates the represented audio signal using the one or more packets from the first audio stream and the corresponding packet to the missing or delayed packet from the second audio stream.

51. The computer readable medium of claim 49, wherein the destination node is further configured to detect that the corresponding packet to the missing or delayed packet in the first audio stream was not received and improve a perceived quality of the regenerated audio signal by replacing audio from the missing or delayed packet in the first audio stream using an interpolation process selected from the group comprising:

interpolating between packets, filling with white noise, and filling with pink noise.

52. The computer readable medium of claim 49, wherein the one or more packets in the first audio stream comprise a plurality of packets, wherein the duration of at least two packets of the plurality of packets in the first audio stream is different, wherein the one or more packets in the second audio stream comprise a plurality of packets, and wherein the duration of at least two packets of the plurality of packets in the second audio stream is different.

53. A method comprising:

generating, by a common source node, a first audio stream representing an audio signal, wherein the first audio stream is encoded using a first encoding technique, and wherein the first audio stream comprises one or more packets, each with a duration and a timestamp;

generating, by the common source node, a second audio stream representing an audio signal, wherein the second audio stream is encoded using a second encoding technique, and wherein the second audio stream comprises one or more packets, each with a duration and a timestamp, the second encoding technique using less bandwidth than the first encoding technique, the second audio stream having a corresponding packet for each packet in the first audio stream that has the same duration and the same timestamp as the corresponding packet in the first audio stream, wherein the second network comprises a link established over the second network that provides a lower bandwidth to transmit the second audio stream when compared to a link established over the first network to transmit the first audio stream;

sending, by the common source node, the first audio stream on the first network; and

sending, by the common source node, the second audio stream on the second network.

54. The method of claim 53, wherein a common destination node receives the first and second audio streams;

detects a missing or delayed packet in the first audio stream; and

regenerates the represented audio signal using the one or more packets from the first audio stream and the corresponding packet to the missing or delayed packet from the second audio stream.

55. The method of claim 53, wherein the destination node is further configured to detect that the corresponding packet to the missing or delayed packet in the first audio stream was not received and improve a perceived quality of the regenerated audio signal by replacing audio from the missing or delayed packet in the first audio stream using an interpolation process selected from the group comprising:

interpolating between packets, filling with white noise, and filling with pink noise.

56. The method of claim 53, wherein the one or more packets in the first audio stream comprise a plurality of packets, wherein the duration of at least two packets of the plurality of packets in the first audio stream is different, wherein the one or more packets in the second audio stream comprise a plurality of packets, and wherein the duration of at least two packets of the plurality of packets in the second audio stream is different.