Audio codec reselection for increased port density

• US 6,445,697 B1
• Filed: 09/25/1998
• Issued: 09/03/2002
• Est. Priority Date: 09/25/1998
• Status: Expired due to Term

• Current Assignee: Cisco Technology, Inc. (Cisco Systems, Inc.)
• Original Assignee: Cisco Technology, Inc. (Cisco Systems, Inc.)
• Inventors: Fenton, James L.
• Primary Examiner(s): Chin, Wellington
• Assistant Examiner(s): Duong, Frank
• Application Number:
  US09/161,111
• Time in Patent Office: 1,439 Days
• Field of Search: 370/351, 370/352, 370/353, 370/354, 370/355, 370/356, 370/357, 370/401, 370/431, 370/437, 370/442, 370/463, 370/464, 370/465, 370/468, 370/477, 709/104, 709/226
• US Class Current: 370/357
• CPC Class Codes: H04M 7/0072 Speech codec negotiation H04L65/1069 takes precedence
• CPC Class Codes: H04M 7/125 Details of gateway equipment

Abstract

An Internet telephony gateway and method for operating a gateway are disclosed. The gateway is designed with a port to support a predefined maximum number of audio data channels. The gateway contains sufficient processing throughput to operate a first, high quality audio codec on a subset of the channels. However, this throughput is sufficient to operate a second, lower quality audio codec on a greater number of the channels, preferably all of them. The first and second codecs are designed to produce compressed audio data streams that are interoperably decompressable.

In operation, the gateway host processor assigns new calls to either the first or second codec, depending on the current traffic being handled by the gateway. If new calls would result in the gateway'"'"'s processing throughput being exceeded, the host processor may reassign a channel from the first codec to the second codec in order to create processing headroom for the addition of a new channel. Because the codecs are interoperably decompressable, no renegotiation need occur with the far end of the communication channel when a codec is reassigned. This gateway offers the potential for high-quality communication over the maximum number of channels possible, with a natural degradation as the gateway reaches its full channel capacity, using modest processing resources.

Claims

1. A packet-switched network telephony gateway comprising:

a port for sending audio data streams to and receiving audio data streams from a first network, said port having a maximum audio data stream capacity of at least two data streams;

a packet switched network interface for formatting the audio data streams into packets for sending over a packet switched network and unformatting packets received from the packet switched network back into the audio data streams;

a host processing resource that monitors the number of audio data streams currently handled by the gateway from the first network and the packet switched network and dynamically associates audio data streams received through said port with a codec selected from a set consisting of a first codec and at least one alternate codec that produce interoperably decompressable data streams; and

a codec processing resource that compresses the audio data streams using the codec currently selected by the host processing resource for that audio data stream, the packet switched network interface packetizing and unpacketizing the compressed audio data streams.

2. The gateway of claim 1, wherein said first network is a circuit-switched network.

3. The gateway of claim 1, wherein said codec processing resource has throughput sufficient to perform real-time data compression using said first codec on only a subset of the maximum audio data stream capacity of said port, but sufficient to perform real-time data compression using said at least one alternate codec on the entire maximum audio data stream capacity of said port.

4. The gateway of claim 1, wherein said host processing resource changes the codec associated with any particular audio data stream depending upon a number of times the host processing resource has previously changed the codec for the audio data stream.

5. The gateway of claim 1, wherein said host processing resource changes the codec associated with any particular audio data stream depending upon whether that particular audio data stream has any current voice activity.

6. The gateway of claim 1, wherein said host processing resource comprises a first microprocessor and said codec processing resource comprises a second microprocessor connected to and controlled by said first microprocessor.

7. The gateway of claim 6, wherein said codec processing resource further comprises a third microprocessor connected to and controlled by said first microprocessor, and wherein said second and third microprocessors receive audio data streams from said port.

8. The gateway of claim 7, wherein said second and third microprocessors can each access any audio data stream received from said port.

9. The gateway of claim 1, wherein said host processing resource changes the codec associated with any particular audio data stream according to a quality of service associated with the packet switched network.

10. The gateway of claim 1, wherein said first codec and said at least one alternate codec both comprise linear predictive audio data compressors operating at a common output data rate, and wherein said first codec requires at least 50% greater average processing time than said second codec.

11. The gateway of claim 10, wherein said audio data compressors both comprise conjugate structure algebraic code excited linear predictive audio data compressors.

12. A packet-switched network telephony gateway comprising:

a port for sending audio data streams to and receiving audio data streams from a circuit-switched network, said port having a maximum audio data stream capacity of at least two data streams;

at least two digital signal processors;

a time-division-multiplexed data bus connected between said port and said digital signal processors, allowing any one of said digital signal processors to access any audio data stream received through said port;

a packet switched network interface for packetizing and depacketing the data streams for communicating over a packet switched network; and

a host processor that monitors the number of audio data streams handled by the gateway, associates each new audio data stream received through said port with one of said digital signal processors for compression, and dynamically associates each audio data stream received through said port with a codec selected from the set consisting of a first codec and a second codec that produce interoperably decompressable data streams according to a number of active audio channels established over the packet switched network.

13. The gateway of claim 12, wherein said at least two digital signal processors have a combined throughput sufficient to perform real-time data compression using said first codec on only a subset of the maximum audio data stream capacity of said port, but sufficient to perform real-time data compression using said at least one alternate codec on the entire maximum audio data stream capacity of said port.

14. The gateway of claim 12, wherein said host processor changes the codec associated with any particular audio data stream according to a number of times the host processor has previously changed the codec associated with the audio data stream.

15. The gateway of claim 12 wherein the host processor changes the codec associated with any particular audio data stream depending upon whether that particular audio data stream has any current voice activity.

16. The gateway of claim 12 wherein the host processor changes the codec associated with any particular audio data stream according to a quality of service associated with the packet switched network.

17. A system comprising:

means for sending audio data streams to and receiving audio data streams from a packet switched network through a port having a maximum audio data stream capacity of at least two data streams;

means for monitoring the number of audio data streams currently handled by the port and dynamically associating audio data streams received through said port with a codec selected from a set consisting of a first codec and at least one alternate codec that produce interoperably decompressable data streams according to the number of audio data streams on the packet switched network handled by the port;

means for compressing each audio data stream using the codec currently selected for that audio data stream; and

means for packetizing and depacketing the compressed audio data stream.

18. The system of claim 17 including means for changing the codec associated with any particular audio data stream according to a number of times the host processing resource has previously changed the codec for the audio data stream.

19. The system of claim 17 including means for changing the codec associated with any particular audio data stream depending upon whether that particular audio data stream has any current voice activity.

20. The system of claim 17 including means for changing the codec associated with any particular audio data stream according to a quality of service associated with the packet switched network.

21. An electronic storage medium containing software used for processing audio data streams, comprising:

code for sending audio data streams to and receiving audio data streams from a packet switched network through a port having a maximum audio data stream capacity of at least two data streams;

code for monitoring the number of audio data streams currently handled by the port and dynamically associating audio data streams received through said port with a codec selected from a set consisting of a first codec and at least one alternate codec that produce interoperably decompressable data streams according to the number of audio data streams on the packet switched network handled by the port;

code for compressing each audio data stream using the codec currently selected for that audio data stream; and

code for packetizing and depacketing the compressed audio data stream.

22. The electronic storage medium of claim 21 including code for changing the codec associated with any particular audio data stream according to a number of times the host processing resource has previously changed the codec for the audio data stream.

23. The electronic storage medium of claim 21 including code for changing the codec associated with any particular audio data stream depending upon whether that particular audio data stream has any current voice activity.

24. The electronic storage medium of claim 21 including code for changing the codec associated with any particular audio data stream according to a quality of service associated with the packet switched network.