Internet telephone system with dynamically varying codec

• US 6,356,545 B1
• Filed: 12/12/1997
• Issued: 03/12/2002
• Est. Priority Date: 08/08/1997
• Status: Expired due to Term

• Current Assignee: Advent IP LLC
• Original Assignee: Clarent Corporation
• Inventors: Vargo, Mike, Chang, Jerry
• Primary Examiner(s): YAO, KWANG BIN
• Application Number:
  US08/989,361
• Time in Patent Office: 1,551 Days
• Field of Search: 370/352, 370/353, 370/354, 370/355, 370/356, 370/465, 370/468, 370/474, 370/477, 370/401, 704/500, 704/501, 704/502, 704/503, 704/504
• US Class Current: 370/355
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: H03M 13/03 Error detection or forward error correction by redundancy in data representation, i.e. code words containing more digits than the source words
• CPC Class Codes: H03M 13/35 Unequal or adaptive error protection, e.g. by providing a different level of protection according to significance of source information or by adapting the coding according to the change of transmission channel characteristics
• CPC Class Codes: H04L 1/0006 by adapting the transmission format
• CPC Class Codes: H04L 1/0009 by adapting the channel coding H04L1/1812 takes precedence
• CPC Class Codes: H04L 1/0014 by adapting the source coding
• CPC Class Codes: H04L 65/103 in the network
• CPC Class Codes: H04L 65/765 intermediate
• CPC Class Codes: H04L 65/80 Responding to QoS
• CPC Class Codes: H04M 3/12 Marking faulty circuits "busy"; Enabling equipment to disengage itself from faulty circuits ; Using redundant circuits; Response of a circuit, apparatus or system to an error
• CPC Class Codes: H04M 7/1275 Methods and means to improve the telephone service quality, e.g. reservation, prioritisation or admission control

Abstract

The invention is concerned with improvements in full duplex Internet telephone systems with a system architecture having low latency and permitting voice communication with telephone to telephone or PC to telephone connections. The architecture permits dynamic packet-to-packet change in codec to adjust for Internet conditions. The voice port creates self-describing packet conditions so that the higher level software of the system is independent of codec selection. In addition to adjusting the codec, the voice port has the capability of dynamically and concurrently selecting other factors such as the level of error correction redundancy, the packet size and packet bundling on a packet-to-packet basis. The invention further includes a technique to eliminate dead air spaces in the voice data transmission stream by speeding up or slowing down the data rate in the buffer while maintaining a constant pitch of speech.

Claims

1. A software method of choosing from a plurality of codecs in an Internet telephone system, said method comprising the steps of:

receiving a plurality of self-describing data packets in a voice data stream on a receiving end;

acquiring a voice quality measurement from said self-describing data packets received at said receiving end; and

dynamically changing codec algorithms in response to said voice quality measurement on a packet-to-packet basis for each packet in said plurality of self-describing data packets for optimizing the voice quality of the information contained in each said packet.

2. The software method of claim 1, further comprising:

varying the length of said packets.

3. The software method of claim 1, further comprising:

applying data redundancy to said packets.

4. The software method of claim 1, further comprising:

varying the bundling of said packets.

5. The software method of claim 1, further comprising the steps of:

detecting a quantity of voice data waiting in a voice input buffer;

regulating the rate of removal of said voice data from said buffer based upon said quantity from a first speed to a second speed; and

maintaining a constant pitch for said voice data as heard by a caller as the rate changes from said first speed to said second speed.

6. The software method of claim 5, wherein said regulating step further comprises:

slowing down the rate of removal of said voice data for low quantities; and

speeding up the rate of removal for high quantities.

7. A software method of choosing from a plurality of codecs in an Internet telephone system, said method comprising the steps of:

sending a plurality of self-describing data packets in a voice data stream on a sending end;

acquiring a voice quality measurement from said self-describing data packets received at a receiving end; and

dynamically changing codec algorithms in response to said voice quality measurement on a packet-to-packet basis for each packet in said plurality of self-describing data packets for optimizing the voice quality of the information contained in each said packet.

8. The software method of claim 7, further comprising:

varying the length of said packets.

9. The software method of claim 7, further comprising:

applying data redundancy of said packets.

10. The software method of claim 7, further comprising:

varying the bundling of said data packets.

11. A software system for choosing from a plurality of codecs in an Internet telephone system, comprising:

a gateway server for receiving a plurality of self-describing data packets in a voice data stream on a receiving end; and

a voice port in said gateway server for acquiring a voice quality measurement from said self-describing data packets received by said gateway server, and

dynamically changing codec algorithms in response to said voice quality measurement on a packet-to-packet basis for each packet in said plurality of self-describing data packets for optimizing the voice quality of the information contained in each said packet.

12. The software system of claim 11, wherein:

said voice port varies the length of said packets.

13. The software system of claim 11, wherein:

said voice port applies data redundancy to said packets.

14. The software system of claim 11, wherein:

said voice port varies the bundling of said packets.

15. The software system of claim 11, further comprising:

a software utility for detecting a quantity of voice data waiting in a voice input buffer;

said utility regulates the rate of removal of said voice data from said buffer based upon said quantity from a first speed to a second speed; and

said utility maintains a constant pitch for said voice data as heard by a caller as the rate changes from said first speed to said second speed.

16. The software system of claim 15, wherein:

said utility slows down the rate of removal of said voice data for low quantities of said data in said buffer and speeds up the rate of removal for high quantities.

17. A software system for choosing from a plurality of codecs in an Internet telephone system, comprising:

a gateway server for receiving a plurality of self-describing data packets in a voice data stream on a receiving end; and

a voice port in said gateway server for

acquiring a voice quality measurement from said self-describing data packets received by said gateway server,

comparing said voice quality measurement to a numerical baseline, and

dynamically changing codec algorithms in response to said comparison of said voice quality measurement to said numerical baseline on a packet-to-packet basis for each packet in said plurality of self-describing data packets for optimizing the voice quality of the information contained in each said packet.

18. The software system of claim 17, wherein:

said voice port varies the length of said packets.

19. The software system of claim 17, wherein:

said voice port applies data redundancy to said packets.

20. The software system of claim 17, wherein:

said voice port varies the bundling of said packets.

21. The software system of claim 17, wherein said codec algorithm is changed if said voice quality measurement is less than said numerical baseline and said codec algorithm is not changed if said measurement is greater than or equal to said numerical baseline.

22. The software system of claim 17, wherein said codec algorithm is changed if said voice quality measurement is greater than or equal to said numerical baseline and said codec algorithm is not changed if said voice quality measurement is less than said baseline.

23. A software system for choosing from a plurality of codecs in an Internet telephone system, comprising:

means for receiving a plurality of self-describing data packets in a voice data stream on a receiving end;

means for acquiring a voice quality measurement from said self-describing data packets received at said receiving end;

means for comparing said voice quality measurement to a numerical baseline; and

means for dynamically changing codec algorithms in response to said comparison of said voice quality measurement to said numerical baseline on a packet-to-packet basis for each packet in said plurality of self-describing packets for optimizing the voice quality of the information contained in each said packet.

24. The software system of claim 23, further comprising:

means for dynamically varying the length of said packets.

25. The software system of claim 23, further comprising:

means for applying data redundancy to said packets.

26. The software system of claim 23, further comprising:

means for varying the bundling of said packets.