CA2452146A1 - Method and system for providing media services - Google Patents

Method and system for providing media services Download PDF

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Publication number
CA2452146A1
CA2452146A1 CA002452146A CA2452146A CA2452146A1 CA 2452146 A1 CA2452146 A1 CA 2452146A1 CA 002452146 A CA002452146 A CA 002452146A CA 2452146 A CA2452146 A CA 2452146A CA 2452146 A1 CA2452146 A1 CA 2452146A1
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Prior art keywords
audio
packets
egress
packet
stream
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Granted
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CA002452146A
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French (fr)
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CA2452146C (en
Inventor
Arthur Irvin Laursen
David Israel
Thomas Mcknight
Serkan Recep Dost
Donald A. Stanwyck
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Movius Interactive Corp
Original Assignee
Ip Unity
Arthur Irvin Laursen
David Israel
Thomas Mcknight
Serkan Recep Dost
Donald A. Stanwyck
Movius Interactive Corporation
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Priority claimed from US09/893,743 external-priority patent/US7161939B2/en
Application filed by Ip Unity, Arthur Irvin Laursen, David Israel, Thomas Mcknight, Serkan Recep Dost, Donald A. Stanwyck, Movius Interactive Corporation filed Critical Ip Unity
Priority to CA2751084A priority Critical patent/CA2751084A1/en
Publication of CA2452146A1 publication Critical patent/CA2452146A1/en
Application granted granted Critical
Publication of CA2452146C publication Critical patent/CA2452146C/en
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L49/00—Packet switching elements
    • H04L49/30—Peripheral units, e.g. input or output ports
    • H04L49/3081—ATM peripheral units, e.g. policing, insertion or extraction
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L61/00—Network arrangements, protocols or services for addressing or naming
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L61/00—Network arrangements, protocols or services for addressing or naming
    • H04L61/30—Managing network names, e.g. use of aliases or nicknames
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L61/00—Network arrangements, protocols or services for addressing or naming
    • H04L61/45—Network directories; Name-to-address mapping
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40—Support for services or applications
    • H04L65/403—Arrangements for multi-party communication, e.g. for conferences
    • H04L65/4038—Arrangements for multi-party communication, e.g. for conferences with floor control
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/487—Arrangements for providing information services, e.g. recorded voice services or time announcements
    • H04M3/493—Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals
    • H04M3/4938—Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals comprising a voice browser which renders and interprets, e.g. VoiceXML
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H04M3/562—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities where the conference facilities are distributed
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q11/00—Selecting arrangements for multiplex systems
    • H04Q11/04—Selecting arrangements for multiplex systems for time-division multiplexing
    • H04Q11/0428—Integrated services digital network, i.e. systems for transmission of different types of digitised signals, e.g. speech, data, telecentral, television signals
    • H04Q11/0478—Provisions for broadband connections
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/02—Details
    • H04L12/16—Arrangements for providing special services to substations
    • H04L12/18—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/54—Store-and-forward switching systems 
    • H04L12/56—Packet switching systems
    • H04L12/5601—Transfer mode dependent, e.g. ATM
    • H04L2012/5638—Services, e.g. multimedia, GOS, QOS
    • H04L2012/5665—Interaction of ATM with other protocols
    • H04L2012/5667—IP over ATM
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/54—Store-and-forward switching systems 
    • H04L12/56—Packet switching systems
    • H04L12/5601—Transfer mode dependent, e.g. ATM
    • H04L2012/5638—Services, e.g. multimedia, GOS, QOS
    • H04L2012/5671—Support of voice

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Multimedia (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Telephonic Communication Services (AREA)

Abstract

The present invention provides a method and system for providing media services in Voice over IP telephony. A switch is coupled between one or more audio sources and a network interface controller. The switch can be a packet switch or a cell switch (304). The present invention further provides a method and system for distributed conference bridge processing in Voice over IP
telephony. A distributed conference bridge multi-casts mixed audio content of a conference call in a way that reduces replication work at the mixing device.
The present invention also provides a method and system for noiselessly switching between independent audio streams. Such noiseless switching preserves valid RTP information at the time of switch over.

Claims (76)

1. A media platform for providing media services over a network, comprising:
a resource manager that manages resources used to support the media services; and an audio processing platform that manages calls and the media services provided in the calls; said audio processing platform including:
a network interface having a set of packet processors that process packets of audio data entering and exiting the media platform in the calls being handled;
a set of audio processors that process the audio data according to the media services provided in the calls; and a switch that switches packets of audio data sent between the audio processors and packet processors.
2. The media platform according to claim 1, wherein said audio processing platform further comprises a call control and audio feature manager that controls resources and media services provided in calls processed by said audio processors.
3. The media platform according to claim 2, wherein said call control and audio feature manager includes:
a call signaling manager;
system manager;
connection manager; and feature controller.
4. The media platform according to claim 2, wherein said audio processing platform comprises a shelf controller card.
5. The media platform according to claim 1, further comprising:
a set of ports coupled to the network; and wherein said network interface further comprises, for each packet processor, a respective controller and forwarding information table.
6. The media platform according to claim 1, wherein said switch comprises a packet switch.
7. The media platform according to claim 1, further comprising a cell layer that combines the packets of audio data into cells of audio, and wherein said switch comprises a cell switch that switches said cells.
8. The media platform according to claim 1, wherein each audio processor comprises a digital signal processor.
9. The media platform according to claim 1, wherein each audio processor comprises a plurality of card processors coupled to a plurality of digital signal processors.
10. The media platform according to claim 1, wherein for at least one ingress audio stream, each packet processor receives IP packets with RTP information from the network and converts the IP packets to internal packets, each internal packet having a payload and header.
11. The media platform according to claim 10, wherein each audio processor processes internal packets.
12. The media platform according to claim 1, wherein for egress audio streams, each packet processor receives internal packets and generates IP packets with RTP
information to be sent over the network.
13. A media platform for providing media services over a network, comprising:
means for managing resources used to support the media services;
means for interfacing with a network, said interface means including means for processing packets of audio data entering and exiting the media platform in calls being handled;
means for processing the audio data according to the media services provided in the calls; and means for switching packets of audio data sent between the audio processors and packet processors.
14. A scalable audio processing platform that manages voice over the Internet calls and media services provided in the calls, the platform including:
a network interface having a set of packet processors that process packets of audio data entering and exiting the platform in the calls being handled;
a set of audio processors that process the audio data according to the media services provided in the calls; and a switch coupled between said network interface and said set of audio processors that switches packets of audio data.
15. A method for providing media services over a network, comprising:
managing resources used to support at least one media service provided to voice over the Internet calls;
processing IP packets of audio data in ingress audio streams and egress audio streams in the calls being handled including converting IP packets to internal packets in ingress audio streams and converting internal packets to IP packets in egress audio streams;
switching the internal packets of audio data in ingress audio streams and egress audio streams in the calls being handled; and processing the internal packets of audio data in ingress audio streams and egress audio streams to provide at least one media service in the calls.
16. A method for processing audio in a conference call among participants, comprising:
(a) generating a fully mixed audio stream of packets, each packet having a packet header and payload;
(b) generating a set of partially mixed audio streams of packets, each packet having a packet header and payload;
(c) multicasting each packet in the fully mixed audio stream and the set of partially mixed audio streams; and (d) determining which multicasted packets to forward based on packet header information in the respective packets.
17. The method of claim 16, further comprising, prior to said steps (a) and (b):
initiating the conference call among the participants; and storing conference identifier information (CID) and network address information associated with each participant in the initiated conference call.
18. The method of claim 17, further comprising:
assigning switched virtual circuits (SVCs) to respective participants in the conference call; wherein said storing step further includes storing said CID
and network address information such that said CID and network address information for each participant can be retrieved based on a respective assigned SVC.
19. The method of claim 16, further comprising:
monitoring energy in inbound audio streams of the participants; and determining a number of active speakers based on the monitored energy.
20. The method of claim 19, wherein said generating steps (a) and (b) generate packet headers having active speaker information based on the determined number of active speakers, and said determining step (d) determines which multicasted packets to forward to participants based on the active speaker information in the respective packet headers.
21. The method of claim 20, wherein the active speaker information comprises TAS and IAS fields, and wherein said generating step (a) generates a fully mixed audio stream of packets having packet headers which include TAS
and IAS fields, and said generating step (b) generates a set of partially mixed audio streams of packets having packet headers which include TAS and IAS fields.
22. The method of claim 21, wherein said determining step (d) determines which multicasted packets to forward to participants based on the information in the TAS and IAS fields in the respective packet headers.
23. The method of claim 21, wherein said determining step (d) further comprises for each packet being processed at a SVC the steps of:
obtaining a CID value for the SVC;
determining whether the obtained CID value matches any CID value in the TAS field of the packet, and if a match exits, determining whether the obtained CID value matches any CID value in the IAS field of the packet, whereby, the packet is discarded when a match exists between the obtained CID value and any CID value in the TAS field and a match exists between the obtained CID value and any CID value in the IAS field.
24. The method of claim 23, further comprising, when a match does not exist between the obtained CID value and any CID value in the TAS field of a packet, comparing the TAS and IAS fields, whereby, when the compared fields are identical the packet can be converted to a network packet, otherwise when the compared fields are not identical the packet can be discarded.
25. The method of claim 21, wherein the packet header further includes sequence information, and wherein said generating step (a) generates a fully mixed audio stream of packets having packet headers which include the sequence information, and said generating step (b) generates a set of partially mixed audio streams of packets having packet headers which include the sequence information.
26. The method of claim 16, wherein:
said generating step (a) generates a fully mixed audio stream of packets, each packet having a packet header and payload, wherein the payload includes mixed audio from at least three active speakers; and said generating step (b) generates a set of partially mixed audio streams of packets, each packet having a packet header and payload, wherein for each partially mixed audio stream of packets the payload includes mixed audio from the at least three active speakers minus the audio of a respective recipient active speaker.
27. The method of claim 16, further comprising:
processing packets determined to be forwarded in said determining step (d) into network packets having network addresses of participants in the conference call; and sending the network packets to the participants.
28. The method of claim 16, further comprising mixing audio received over a network from participants in the conference call who are active speakers.
29. The method of claim 16, further comprising mixing audio received from an internal audio source and audio received over a network from participants in the conference call who are active speakers.
30. A conference bridge that processes audio in a conference call among participants, comprising:
an audio source that generates a fully mixed audio stream of packets and a set of partially mixed audio streams of packets, each packet having a packet header and payload;
a switch; and a network interface controller, wherein said switch is coupled between said network interface controller and said audio source and said switch further comprises a multicaster; and wherein said multicaster multicasts each packet in the fully mixed audio stream and the set of partially mixed audio streams to the network interface controller, and said network interface controller determines which multicasted packets to forward based on packet header information in the respective packets.
31. The conference bridge of claim 30, further comprising a conference call agent that initiates a conference call among the participants.
32. The conference bridge of claim 31, further comprising a storage device that stores conference identifier information (CID) and network address information associated with each participant in the established conference call.
33. The conference bridge of claim 32, wherein said storage device comprises a look-up table.
34. The conference bridge of claim 32, wherein said network interface controller assigns switched virtual circuits (SVCs) to respective participants in the initiated conference call, and said storage device stores said CID and network address information such that said CID and network address information for each participant can be retrieved based on a respective assigned SVC.
35. The conference bridge of claim 30, wherein said audio source monitors energy in inbound audio streams of the participants, and determines a number of active speakers based on the monitored energy.
36. The conference bridge of claim 35, wherein said packet headers generated by said audio source have active speaker information based on the determined number of active speakers, and said network interface controller determines which multicasted packets to forward to participants based on the active speaker information in the respective packet headers.
37. The conference bridge of claim 36, wherein the active speaker information comprises TAS and IAS fields, and said network interface controller determines which multicasted packets to forward to participants based on the information in said TAS and IAS fields in the respective packet headers.
38. The conference bridge of claim 37, wherein said packet headers generated by said audio source further include sequence information.
39. The conference bridge of claim 30, wherein said fully mixed audio stream of packets have payloads which include mixed audio from at least three active speakers, and each partially mixed audio stream of packets in said set have payloads which include mixed audio from the at least three active speakers minus the audio of a respective recipient active speaker.
40. The conference bridge of claim 30, further comprising:
a packet processor that processes packets determined to be forwarded into network packets having network addresses of participants in the conference call.
41. The conference bridge of claim 30, wherein said audio source mixes audio received over a network from participants in the conference call who are active speakers.
42. The conference bridge of claim 30, wherein said audio source mixes audio received from an internal audio source and audio received over a network from participants in the conference call who are active speakers.
43. A system for processing audio in a conference call among participants, comprising:
(a) means for generating a fully mixed audio stream of packets, each packet having a packet header and payload;
(b) means for generating a set of partially mixed audio streams of packets, each packet having a packet header and payload;
(c) means for multicasting each packet in the fully mixed audio stream and the set of partially mixed audio streams; and (d) means for determining which multicasted packets to forward based on packet header information in the respective packets.
44. A media server for use in a VOID network, comprising:
a distributed conference bridge that processes audio in a conference call among participants, said distributed conference bridge comprising:
an audio source that generates a fully mixed audio stream of packets and a set of partially mixed audio streams of packets, each packet having a packet header and payload;
a switch; and a network interface controller, wherein said switch is coupled between said network interface controller and said audio source and said switch further comprises a multicaster; and wherein said multicaster multicasts each packet in the fully mixed audio stream and the set of partially mixed audio streams to the network interface controller, and said network interface controller determines which multicasted packets to forward based on packet header information in the respective packets.
45. A method for noiselessly switching audio provided on an egress audio channel over a network, comprising:
(a) generating a first audio stream of egress packets for the egress audio channel, wherein each egress packet includes a payload carrying audio and control header information;
(b) switching and delivering said first audio stream to a first network interface controller associated with the egress audio channel;
(c) generating a second audio stream of egress packets, wherein each egress packet includes a payload carrying audio and control header information;
(d) switching and delivering said second audio stream to the first network interface controller associated with the egress audio channel;
and (e) evaluating the relative priority of the first and second audio streams based on priority information in the control header information of the egress packets to determine which of the first and second audio streams is a higher priority audio stream to transmit on the egress audio channel over the network.
46. The method of claim 45, further comprising:
packetizing the higher priority audio stream to create an output egress audio stream of packets with synchronized header information; and transmitting the output egress audio stream of packets on the egress audio channel over the network.
47. The method of claim 45, further comprising:
packetizing the lower priority audio stream to create an output egress audio stream of packets with synchronized header information, whereby, synchronized header information is noiselessly preserved in IP packets transmitted on the egress audio channel over the network for audio from both of the first and second audio streams.
48. The method of claim 45, further comprising:
converting the first audio stream of egress packets into first cells; and converting the second audio stream of egress packets into second cells;
wherein said switching step (b) comprises switching the converted first cells to a SVC associated with the egress audio channel, and said switching step (d) comprises switching the converted second cells to the SVC associated with the egress audio channel.
49. The method of claim 46, wherein said synchronized header information comprises valid RTP information.
50. The method of claim 45, further comprising:
(f) determining synchronized RTP header information for each of the first and second audio streams prior to transmitting IP packets containing audio payloads of the respective first and second audio streams on the egress audio channel over the network.
51. A method for noiselessly switching audio from a second audio source to an egress audio channel already carrying audio from a first audio source, comprising:
generating an audio stream of egress packets at the second audio source;
converting the audio stream of egress packets into cells;

switching the converted cells to a switched virtual circuit (SVC) associated with the egress audio channel;
converting the switched cells back to the audio stream of egress packets;
packetizing the audio stream to create an output egress audio stream of packets with synchronized header information; and transmitting the output egress audio stream of packets on the egress audio channel over a network in place of the audio from the first audio source.
52. The method of claim 51, wherein said generating step generates an audio stream of egress packets at the second audio source in response to a call event.
53. The method of claim 51, wherein said generating step generates an audio stream of egress packets at the second audio source in response to a call event, the audio stream of egress packets including a type of audio selected from at least one of voice, music, tones, or sound.
54. The method of claim 53, further comprising generating the call event based on at least one of the following conditions: an emergency condition, a call signaling condition, a call event based on callee or caller information, or a request for audio information.
55. The method of claim 53, further comprising generating the call event based on a request for audio information, wherein the request for audio information includes at least one of a request for advertisements, news, sports, financial, music or other audio content.
56. A method for introducing noiseless switch over audio for voice over the Internet (VOID) telephone calls, comprising:

establishing a VOID telephone call between a destination device and a media server;
setting priority information for a first audio source;
delivering a first audio stream of egress packets that includes the set priority information;
determining a call status with respect to availability of receiving noiseless switch over audio; and processing a call event that includes noiseless switch over audio when the call status determining step indicates the established VOID telephone call is a candidate for receiving noiseless switch over audio.
57. The method of claim 56, wherein said processing step includes:
determining priority information for the noiseless switch over audio; and transmitting the noiseless switch over audio in an output audio stream of packets in the established VOID telephone call when the determined priority information for the noiseless switch over audio is greater than the set priority information of the first audio stream.
58. The method of claim 57, further comprising:
generating a second audio stream of egress packets at a second audio source, the audio stream having the noiseless switch over audio in payloads;
converting the second audio stream of egress packets into cells;
switching the converted cells to a SVC associated with an egress audio channel of the established VOID telephone call;
converting the switched cells back to the second audio stream of egress packets;
packetizing the second audio stream with synchronized header information to create the output audio stream of packets in the established VOID telephone call; and transmitting the output audio stream of packets on the egress audio channel in the established VOID telephone call over a network in place of the audio from the first audio source.
59. A system for noiselessly switching audio provided on an egress audio channel over a network, comprising:
first and second audio sources;
a switch coupled to said first and second audio sources; and a network interface controller coupled to said switch, wherein said first audio source generates a first audio stream of egress packets for the egress audio channel, wherein each egress packet includes a payload carrying audio and control header information, said second audio source generates a second audio stream of egress packets, wherein each egress packet includes a payload carrying audio and control header information, and said switch switches and delivers said first and second audio streams to said network interface controller.
60. The system of claim 59, further comprising:
an egress audio controller coupled to said second audio source, wherein said egress audio controller sends a control signal to said second audio source to initiate the generation of said second audio stream.
61. The system of claim 60, wherein said egress audio controller is further coupled to said first audio source, said switch, and said network interface controller, and wherein said egress audio controller sends a control signal to said first audio source to initiate the generation of said first audio stream when a VOID
telephone call is established, sends a control signal to said switch identifying said network interface controller as being associated with an egress audio output channel associated with the established VOID telephone call, and sends a control signal to said network interface controller as being associated with an egress audio output channel associated with the established VOID telephone call.
62. The system of claim 61, wherein said egress audio controller is further coupled to said first audio source, and wherein said egress audio controller sends control signals to said first and second audio sources to set priority information in said first and second audio streams.
63. The system of claim 59, further comprising at least one packet processor that generates IP packets having synchronized header information and audio payloads, the audio payloads including audio payloads carried in said first and second audio streams.
64. The system of claim 63, wherein said network interface controller dynamically selects which of the IP packets to transmit based on the relative priority of the first and second audio streams; and wherein said switch comprises a packet switch or a cell switch.
65. The system of claim 59, wherein at least one of said first audio source and said second audio source internally generates the audio for said respective first and second audio streams.
66. The system of claim 59, wherein at least one of said first audio source and said second audio source converts audio from an external source to generate the audio for said respective first and second audio streams.
67. A system for noiselessly switching audio from a second audio source to an egress audio channel already carrying audio from a first audio source, comprising:

means for generating an audio stream of egress packets at the second audio source;
means for converting the audio stream of egress packets into cells;
means for switching the converted cells to a SVC associated with the egress audio channel;
means for converting the switched cells back to the audio stream of egress packets;
means for packetizing the audio stream to create an output egress audio stream of packets; and means for transmitting the output egress audio stream of packets on the egress audio channel over a network in place of the audio from the first audio source.
68. A system for introducing noiseless switch over audio for voice over the Internet (VOIP) telephone calls, comprising:
means for establishing a VOIP telephone call between a destination device and a media server;
means for setting priority information for a first audio source;
means for delivering a first audio stream of egress packets that includes the set priority information;
means for determining a call status with respect to availability of receiving noiseless switch over audio; and means for processing a call event that includes noiseless switch over audio when the call status determining step indicates the established VOIP telephone call is a candidate for receiving noiseless switch over audio.
69. The system of claim 68, wherein said processing means includes:
means for determining priority information for the noiseless switch over audio; and means for transmitting the noiseless switch over audio in an output audio stream of packets having synchronized header information in the established VOIP telephone call when the determined priority information for the switch over audio is greater than the set priority information of the first audio stream.
70. The system of claim 69, further comprising:
means for generating a second audio stream of egress packets at a second audio source, the audio stream having the noiseless switch over audio in payloads;
means for converting the second audio stream of egress packets into cells;
means for switching the converted cells to a SVC associated with an egress audio channel of the established VOIP telephone call;
means for converting the switched cells back to the second audio stream of egress packets;
means for packetizing the second audio stream to create the output audio stream of packets in the established VOIP telephone call; and means for transmitting the output audio stream of packets on the egress audio channel in the established VOIP telephone call over a network in place of the audio from the first audio source.
71. A method for introducing noiseless switch over audio for voice over the Internet (VOIP) telephone calls, comprising:
establishing a VOIP telephone call; and transmitting noiseless switch over audio in an output audio stream of packets with synchronized header information in the established VOIP telephone call.
72. A method for noiseless switching between audio sources in a VOIP
network, comprising:
(A) selecting one audio source;
(B) transmitting audio from the selected one audio source in an output audio stream of packets with synchronized header information on an egress audio channel to a destination device;

(C) selecting another audio source; and (D) transmitting audio from the selected another audio source in an output audio stream of packets with synchronized header information on the same egress audio channel to the destination device.
73. The method of claim 72, wherein the another audio source comprises an internal audio source, and further comprising generating audio payloads for the output audio stream of packets prior to said transmitting step (B).
74. The method of claim 72, wherein the another audio source comprises an external audio source, and further comprising extracting audio payloads for the output audio stream of packets from IP packets generated at the external audio source prior to said transmitting step (B).
75. A method, comprising:
(A) transmitting audio from one audio source in an output audio stream of packets with synchronized header information on an egress audio channel to a destination device; and (B) transmitting audio from another independent audio source in an output audio stream of packets with synchronized header information on the same egress audio channel to the destination device, whereby a user at the destination device perceives a noiseless switch over between transmitted audio from independent audio sources in a VOIP network.
76. A system, comprising:
(A) means for transmitting audio from one audio source in an output audio stream of packets with synchronized header information on an egress audio channel to a destination device; and (B) means for transmitting audio from another independent audio source in an output audio stream of packets with synchronized header information on the same egress audio channel to the destination device, whereby a user at the destination device perceives a noiseless switch over between transmitted audio from independent audio sources in a VOIP network.
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US09/930,500 US6847618B2 (en) 2001-06-29 2001-08-16 Method and system for distributed conference bridge processing
US10/122,397 2002-04-16
US10/122,397 US6947417B2 (en) 2001-06-29 2002-04-16 Method and system for providing media services
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WO2003003157A9 (en) 2003-03-20
WO2003003157A2 (en) 2003-01-09
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US6947417B2 (en) 2005-09-20
US20030002481A1 (en) 2003-01-02
JP2004534457A (en) 2004-11-11
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