SYSTEM AND METHOD OF VOICE OVER INTERNET PROTOCOL COMMUNICATION

• US 20090067341A1
• Filed: 10/31/2008
• Published: 03/12/2009
• Est. Priority Date: 03/11/2004
• Status: Active Grant

• Current Assignee: I2Telecom International Inc.
• Original Assignee: I2Telecom International Inc.
• Inventors: Awais, Ali
• Granted Patent: US 8,335,232 B2
• Time in Patent Office: Days
• US Class Current: 370/253
• 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 65/1083 In-session procedures
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04L 65/70 Media network packetisation
• CPC Class Codes: H04L 65/80 Responding to QoS
• CPC Class Codes: H04M 7/0072 Speech codec negotiation H04L65/1069 takes precedence
• CPC Class Codes: H04W 88/181 Transcoding devices; Rate adaptation devices

Abstract

Systems and methods of Voice over Internet Protocol (VOIP) communication are provided. A particular system includes a VoIP endpoint. The VoIP endpoint is adapted to monitor a packet loss measure between the VoIP endpoint and a second VoIP endpoint during a VoIP communication using a first CODEC. The VoIP endpoint is also adapted to negotiate with the second VoIP endpoint to select a substitute CODEC having a nominal data rate that is lower than an in-use nominal data rate of the first CODEC.

Claims

1. A method, comprising:

monitoring a packet loss measure during a Voice over Internet Protocol (VoIP) communication between a first endpoint and a second endpoint, wherein the first endpoint and the second endpoint use a first CODEC to conduct the VoIP communication;

when the monitored packet loss measure exceeds a maximum packet loss threshold and when a current frames per packet value used by the first CODEC satisfies a maximum frames per packet threshold associated with the first CODEC, negotiating use of a substitute CODEC having a substitute nominal data rate that is lower than an in-use nominal data rate of the first CODEC; and

continuing the VoIP communication using the substitute CODEC.

2. The method of claim 1, further comprising determining a starting bandwidth measure to select the first CODEC before beginning to monitor the packet loss measure.

3. The method of claim 2, wherein determining the starting bandwidth measure comprises:

sending a starting probe packet from the first endpoint to a network device;

receiving an echoed starting probe packet from the network device at the first endpoint; and

determining the bandwidth of the path from the first endpoint to the network device.

4. The method of claim 3, wherein the network device is selected from a STUN server, a SIP server, and an echo server.

5. The method of claim 2, further comprising negotiating the first CODEC based on a plurality of CODECs available at the first endpoint, a plurality of CODECs available at the second endpoint, and the starting bandwidth measure.

6. The method of claim 5, wherein negotiating the first CODEC includes determining a bit rate and a frames per packet value to be used by the first endpoint and the second endpoint.

7. The method of claim 5, wherein negotiating the first CODEC comprises:

accessing an optimization database, wherein the optimization database lists a plurality of CODECs and a bandwidth range associated with each of the plurality of CODEC; and

selecting as the first CODEC a CODEC associated with a particular bandwidth range within which the starting bandwidth measure falls.

8. The method of claim 1, further comprising, before negotiating use of the substitute CODEC:

determining a packet loss ratio of a media packet stream of the VoIP communication;

determining whether the packet loss ratio exceeds a maximum packet loss ratio associated with the first CODEC;

when the packet loss ratio exceeds the maximum packet loss ratio, determining the in-use nominal data rate of the first CODEC;

determining whether the first CODEC is associated with an alternate nominal data rate that is lower than the in-use nominal data rate; and

when the first CODEC is associated with an alternate nominal data rate that is lower than the in-use nominal data rate, substituting the alternate nominal data rate for the in-use nominal data rate.

9. The method of claim 8, further comprising:

when the first CODEC is not associated with an alternate nominal data rate that is lower than the in-use nominal data rate, determining whether the current frames per packet value is less than the maximum frames per packet measure associated with the first CODEC; and

when the current frames per packet measure is less than the maximum frames per packet measure associated with the first CODEC, increasing a number of frames per packet used by the first CODEC.

10. The method of claim 1, wherein the packet loss measure includes a ratio of a media packet stream between the first endpoint and the second endpoint.

11. A system, comprising:

a Voice over Internet Protocol (VOIP) endpoint, the VoIP endpoint adapted to;

monitor a packet loss measure between the VoIP endpoint and a second VoIP endpoint during a VoIP communication using a first CODEC; and

negotiate with the second VoIP endpoint to select a substitute CODEC having a nominal data rate that is lower than an in-use nominal data rate of the first CODEC when the monitored packet loss measure exceeds a maximum packet loss threshold and when a frames per packet value used for the VoIP communication satisfies a maximum frames per packet threshold.

12. The system of claim 11, further comprising a telephone coupled to the VoIP endpoint.

13. The system of claim 11, wherein the VoIP endpoint comprises a telephone.

14. The system of claim 11, wherein the substitute CODEC and a frame per packet value for the substitute CODEC are selected from a plurality of available CODEC and frame per packet value combinations, wherein the substitute CODEC and the frame per packet value are selected to provide an enhanced quality of service.

15. The system of claim 11, wherein the VoIP endpoint is further adapted to determine a measurement of available bandwidth between the VoIP endpoint and the second VoIP endpoint, and to adjust a transmission rate of the VoIP communication based on the measurement of available bandwidth.

16. The system of claim 15, wherein the VoIP communication is routed via a network including a plurality of network elements, and wherein the VoIP endpoint determines the measurement of available bandwidth by sending one or more probe packets to one or more of the plurality of network elements.

17. The system of claim 16, wherein the one or more probe packets are sent during a period of silence of the VoIP communication.

18. The system of claim 11, wherein the VoIP endpoint is further adapted to echo a probe packet received from the second VoIP endpoint.

19. The system of claim 11, wherein the VoIP endpoint has access to an optimization database, wherein the optimization database includes at least one frame per packet value for each of a plurality of available CODECs,

20. The system of claim 19, wherein the optimization database further includes a minimum bandwidth for each of the plurality of available CODECs and frame per packet value combinations.

21. The system of claim 19, wherein the optimization database further includes a maximum tolerable packet loss ratio for each of the plurality of available CODECs and frame per packet value combinations.

22. The system of claim 11, wherein negotiating with the second VoIP endpoint to select the substitute CODEC occurs after the monitored packet loss measure has exceeded the maximum packet loss threshold for a predetermined amount of time.

23. The system of claim 11, wherein the VoIP endpoint negotiates with the second VoIP endpoint to return to the first CODEC when conditions are suitable.