Media flow converter for use in real-time delivery transactions

• US 20070116039A1
• Filed: 11/23/2005
• Published: 05/24/2007
• Est. Priority Date: 11/23/2005
• Status: Active Grant

• Current Assignee: International Business Machines Corporation
• Original Assignee: International Business Machines Corporation
• Inventors: Muschett, Brien, Nusbickel, Wendi, Gavagni, Brett
• Granted Patent: US 7,925,769 B2
• Time in Patent Office: Days
• US Class Current: 370/465
• CPC Class Codes: H04L 65/65 Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
• CPC Class Codes: H04L 65/756 adapting media to device capabilities
• CPC Class Codes: H04L 67/62 Establishing a time schedule for servicing the requests

Abstract

A method (200) and a system (100) for coordinated streaming use a single Real Time Protocol (RTP) producer (130) for handling multiple audio services (110). The method can include the steps of assigning (202) a RTP producer to handle multiple audio objects, and maintaining (204) a service for each object in accordance with a delivery schedule. RTP packets can be sent in accordance with the delivery schedule for complying with real-time requirements of a media rendering client thereby providing continuous real-time service delivery. The method can further include determining a wait time and updating the delivery schedule in view of the wait time. In one arrangement, the RTP producer can sleep for a pre-specified interval, and upon wake, prioritizes service delivery based on an audio object'"'"'s wait time.

Claims

1. A media flow converter for use with a server for coordinated streaming using a single Real Time Protocol (RTP) producer for handling multiple audio services comprising:

at least one service application running on said server;

a media converter interfacing with said service application for receiving at least one media stream, wherein said media stream corresponds with at least one service; and

at least one media-rendering client communicatively linked to said media converter for receiving said at least one continuous media stream from said server, wherein said at least one continuous media stream corresponds to said at least one media stream for rendering the at least one said service in real-time.

2. The system of claim 1, wherein the media converter further comprises:

at least one of a plurality of audio objects each supporting an audio channel, wherein each audio object packetizes said media stream into RTP packets on said audio channel; and

a single priority thread for handling said audio objects with respect to a delivery schedule;

wherein said audio objects send RTP packets from said audio channels to at least one media rendering client.

3. The system of claim 2, wherein said delivery schedule complies with real-time requirements of said media rendering client for providing continuous real-time delivery from said server to said media rendering client over said continuous media stream.

4. The system of claim 1, wherein the media converter further includes a native timer that determines a sleep time and a wait time for each audio object.

5. The system of claim 4, wherein said RTP producer updates said delivery schedule on based on a wait time of each audio object, wherein said wait time is the amount of time an audio object has been waiting to send RTP packets.

6. The system of claim 4, wherein said RTP producer sleeps for a pre-specified interval, and upon wake, prioritizes service delivery based on said audio objects'"'"' wait time.

7. The system of claim 1, wherein said at least one service application can be one of a text-to-speech service, an audio processing service, and a music processing service.

8. The system of claim 1, wherein said at least one service application is provided by a WebSphere Voice Server running on said server, wherein said server is a WebSphere Application Server, and wherein WebSphere Voice Server is integrated with said WebSphere Application Server for providing a mix of java transaction based processing and soft-real-time processing for interfacing with said media converter using a J2EE framework.

9. The system of claim 8, wherein WebSphere Voice Server provides speech recognition and synthesis support to at least one said media converter hosting a Web-based VoiceXML application, wherein the support is for at least one real-time continuous media stream connecting said WebSphere Application Server with said Web-based VoiceXML application.

10. The system of claim 8, wherein said single priority thread controls communication between said WebSphere Voice Server and WebSphere Application Server for achieving real-time delivery.

11. A method for coordinated streaming, the method comprising the steps of:

assigning a RTP producer to handle multiple audio objects; and

maintaining a service for each object in accordance with a delivery schedule.

12. The method of claim 11, further comprising sending RTP packets in accordance with said delivery schedule for complying with real-time requirements of a media rendering client thereby providing continuous real-time service delivery.

13. The method of claim 11, further comprising:

determining at least one wait time that an audio object has been waiting to send RTP packets; and

updating said delivery schedule in view of said wait time.

14. The method of claim 11, wherein said RTP producer sleeps for a pre-specified interval, and upon wake, prioritizes service delivery based on said audio objects'"'"' wait time

15. The method of claim 13, further comprising the step of obtaining reference to a native timer which determines a sleep time and wait time for an audio object.

16. In a web-based voice application hosted by a server, a method for packetizing at least one media stream into a continuous media stream, the method comprising:

receiving a media stream on an audio channel, said audio channel having a corresponding audio thread for controlling access to said audio channel;

packetizing said media stream into RTP packets on an audio queue within said audio channel.

17. The method of claim 16, further comprising:

based on a delivery schedule established by a single RTP producer thread, removing RTP packets from said audio queue;

sending RTP packets to at least one media rendering client, wherein the sending provides at least one service application running on said server.

18. The system of claim 16, wherein said at least one service application can be one of a text-to-speech service, an audio processing service, and a music processing service.

19. The system of claim 16, wherein said at least one service application is provided by a WebSphere Voice Server running on said server, wherein said server is a WebSphere Application Server, wherein WebSphere Voice Server is integrated with said WebSphere Application Server for providing a mix of java transaction based processing and soft-real-time processing for interfacing with said media converter using a J2EE framework.

20. In a WebSphere Voice Server application hosted by a WebSphere Application Server, a method for packetizing at least one non-real-time media stream into a continuous real-time media stream for RTP delivery complying with real-time requirements of a media rendering client for providing continuous real-time delivery of said continuous media stream, the method comprising:

receiving audio media from at least one service;

in at least one audio thread, packetizing said audio media to RTP packets;

in said at least one audio thread, placing said RTP packets on a queue in non-real time;

at an RTP producer thread, waking up from a sleep;

at said RTP producer thread, based on a delivery schedule, checking said at least one audio thread for timing information;

removing RTP packets from said queue; and

sending RTP packets to at least one media rendering client,

wherein said delivery schedule complies with real-time requirements of said media rendering client.