System and method for handling media streams

• US 8,582,559 B2
• Filed: 08/02/2007
• Issued: 11/12/2013
• Est. Priority Date: 08/03/2006
• Status: Active Grant

• Current Assignee: Alvaria Cayman
• Original Assignee: Aspect Software Incorporated
• Inventors: Matias, John P., Punreddy, Ajay, Snider, Marc A.
• Primary Examiner(s): Phung, Luat
• Assistant Examiner(s): CHANG, KAI J
• Application Number:
  US11/888,987
• Publication Number:
  US 20080031231A1
• Time in Patent Office: 2,294 Days
• Field of Search: 370352-356, 370/231, 370/230.1, 370/466, 370/471, 370/260, 379/69, 379/88.17, 379/201.01
• US Class Current: 370/352
• CPC Class Codes: H04L 65/103 in the network
• CPC Class Codes: H04L 65/104 in the network
• CPC Class Codes: H04L 65/1043 Gateway controllers, e.g. media gateway control protocol [MGCP] controllers
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04L 65/765 intermediate

Abstract

A mechanism for managing and enhancing communication streams in a VOIP service using SIP is discussed. SIP is used to manage and enhance a communication stream at a centralized switching element in the VoIP environment rather than at the communication stream endpoints. The communication stream set up by SIP is directed through a central switching element under administrative control rather than traveling directly between the stream endpoints. As a result, the media stream may be conditioned with treatments and may be provided with enhanced services at the central switching element without the specific treatments and services being supported at both stream endpoints. The use of the centralized switching element allows full administrative control of the VoIP environment.

Claims

1. A computer-implemented method for managing and enhancing communication streams in a VoIP environment, the method comprising the steps of:

establishing, with a control module that supports a plurality of Session Initiation Protocol (SIP) features, a communication stream between a calling unit and a called unit;

the control module directing the communication stream to pass bi-directionally through at least one relay module that supports the plurality of SIP features;

managing, from the control module, the communication stream passing through the at least one relay module, the managing of the communication stream performed by the control module by sending directives using SIP to the at least one relay module to condition the communications stream at the at least one relay module with at least one feature of the plurality of SIP features which is not supported by at least one of the called unit and the calling unit; and

providing an enhanced service which is not supported by the at least one of the calling unit and the called unit to the communication stream at the at least one relay module using SIP, wherein the enhanced service is at least one of a call recording, an application of global policy information, and an application of an enhanced speech processing technology, and wherein the control module establishes the communications stream, after determining the called unit is available in response to a communications invitation received from the calling unit identifying the called unit, by providing port and codec information of the at least one relay module.

2. The computer-implemented method of claim 1, further comprising:

conditioning the communication stream at the at least one relay module using SIP.

3. The computer-implemented method of claim 2 wherein the communication stream is transcoded to change the communications stream from one format to another as the communication stream passes through the at least one relay module.

4. The computer-implemented method of claim 1 wherein the application of the enhanced speech processing technology provides packet loss concealment for the communication stream.

5. A computer-implemented system for managing and enhancing communication streams in a VoIP environment, comprising:

a first computing device having a calling unit;

a called unit on a second computing device, the second computing device accessible over a network by the first computing device;

a control module that supports a plurality of Session Initiation Protocol (SIP) features, the control module configured to establish a bi-directional communication stream between the calling unit and the called unit;

at least one relay module that supports the plurality of SIP features, the bi-directional communication stream directed through the at least one relay module by the control module, the bi-directional communication stream being managed at the at least one relay module by the control module, the managing of the bi-directional communication stream performed by the control module by sending directives using SIP to the at least one relay module to condition the bi-directional communications streams at the at least one relay module with at least one SIP feature of the plurality of SIP features which is not supported by at least one of the called unit and the calling unit; and

the control module provides an enhanced service which is not supported by the at least one of the calling unit and the called unit to the communication stream at the at least one relay module using SIP, wherein the enhanced service is at least one of a call recording, an application of global policy information, and an application of an enhanced speech processing technology, and wherein the control module establishes the communications stream, after determining the called unit is available in response to a communications invitation received from the calling unit identifying the called unit, by providing port and codec information of the at least one relay module.

6. The computer-implemented system of claim 5 wherein the bi-directional communications stream is conditioned at the at least one relay module using SIP.

7. The computer-implemented system of claim 5 wherein statistics are compiled about the bi-directional communications stream by monitoring the bi-directional communications stream at the relay module.

8. The computer-implemented system of claim 7 wherein the statistics track packet loss in the bi-directional communications stream.

9. The computer-implemented system of claim 7 wherein the statistics track jitter in the bi-directional communications stream.

10. The computer-implemented system of claim 5 wherein the computer-implemented system includes at least two relay modules and the bi-directional communications stream is passed between the at least two relay modules.

11. A non-transitory computer readable medium for use with a computing device, the non-transitory computer readable medium holding computer-executable instructions for managing and enhancing communication streams in a VoIP environment, the instructions comprising:

instructions for establishing, with a control module that supports a plurality of SIP features, a communication stream between a calling unit and a called unit;

instructions for providing at least one relay module that supports the plurality of SIP features, the control module directing the communication stream to pass through the at least one relay module;

instructions for managing, from the control module, the communication stream passing through the at least one relay module, the managing of the communication stream performed by the control module by sending directives using SIP (Session Initiation Protocol) to the at least one relay module to condition the communications stream with at least one feature of the plurality of SIP features which is not supported by at least one of the calling unit and the called unit; and

instructions for providing an enhanced service which is not supported by the at least one of the calling unit and the called unit to the communication stream at the at least one relay module using SIP, wherein the enhanced service is at least one of a call recording, an application of global policy information, and an application of an enhanced speech processing technology, and wherein the control module establishes the communications stream, after determining the called unit is available in response to a communications invitation received from the calling unit identifying the called unit, by providing port and codec information of the at least one relay module.

12. The non-transitory computer readable medium of claim 11, the instructions further comprising:

instructions for conditioning the communication stream at the at least one relay module using SIP.

13. The non-transitory computer readable medium of claim 12 wherein the communication stream is transcoded.

14. The non-transitory computer readable medium of claim 11 wherein the application of global policy information affects packet marking and handling.

15. The non-transitory computer readable medium of claim 11 wherein the application of the enhanced speech processing technology provides packet loss concealment for the communication stream.