Title of Invention	A METHOD AND SYSTEM FOR PROCESSING AUDIO IN A CONFERENCE CALL AMONG PARTICIPANTS
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.

Title of Invention

A METHOD AND SYSTEM FOR PROCESSING AUDIO IN A CONFERENCE CALL AMONG PARTICIPANTS

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.

Full Text	METHOD AND SYSTEM FOR PROCESSING AUDIO IN A CONFERENCE CALL AMONG PARTICIPANTS BACKGROUND OF THE INVENTION Field of the Invention The invention relates generally to audio communication over a network. Background Art Audio has long been carried in telephone calls over networks. Traditional circuit-switched time division multiplexing (TDM) networks including public- switched telephone networks (PSTN) and plain old telephone networks (POTS) were used. These circuit-switched networks establish a circuit across the network for each call. Audio is carried in analog and/or digital form across the circuit in real-time. The emergence of packet-switched networks, such as the local area networks (LANs), and the Internet, now requires that audio be carried digitally in packets. Audio can include but is not limited to voice, music, or other type of audio data. Voice over Internet Protocol systems (also called Voice over IP or VOIP systems) transport the digital audip data belonging to a telephone call in packets over packet-switched networks instead of traditional circuit-switched networks. In one example, a VOIP system forms two or more connections using Transmission Control Protocol/Internet Protocol (TCP/IP) addresses to accomplish a connected telephone call. Devices that connect to a VOIP network must follow standard TCP/IP packet protocols in order to intemperate with other devices within the VOIP network. Examples of such devices are IP phones, integrated access devices, media gateways, and media servers. A media server is often an endpoint in a VOIP telephone call. The media server is responsible for ingress and egress audio streams, that is, audio streams which enter and leave a media server respectively. The type of audio produced by a media server is controlled by the application that corresponds to the telephone call such as voice mail, conference bridge, interactive voice response (IVR), speech recognition, etc. In many applications, the produced audio is not predictable and must vary based on end user responses. Words, sentences, and whole audio segments such as music must be assembled dynamically in real time as they are played out in audio streams. Packet-switched networks, however, can impart delay and jitter in a stream of audio carried in a telephone call. A real-time transport protocol (RTP) is often used to control delays, packet loss and latency in an audio stream played out of a media server. The audio stream can be played out using RTP over a network link to a real-time device (such as a telephone) or a non-real-time device (such as an email client in unified messaging). RTP operates on top of a protocol such as the User Datagram Protocol (UDP) which is part of the IP family. RTP packets include among other things a sequence number and a timestamp. The sequence number allows a destination application using RTP to detect the occurrence of lost packets and to ensure a correct order of packets are presented to a user. The timestamp corresponds to the time at which the packet was assembled. The timestamp allows a destination application to ensure synchronized play-out to a destination user and to calculate delay and jitter. See, D. Collins, Carrier Grade Voice over IP,Mc-Grsw Hill: United States, Copyright 2001, pp. 52-72, the entire book of which is incorporated in its entirety herein by reference. A media server at an endpoint in a VOIP telephone call uses protocols such as RTP to improve communication quality for a single audio stream. Such media servers, however, have been limited to outputting a single audio stream of RTP packets for a given telephone call. A conference call links multiple parties over a network in a common call. Conference calls were originally carried out over a circuit-switched network such as a plain old telephone system (POTS) or public switched telephone network (PSTN). Conference calls are now also carried out over packet-switched networks, such as local area networks (LANs) and the Internet. Indeed, the emergence of voice over the Internet systems (also called Voice over IP or VOIP systems) has increased the demand for conference calls over networks. Conference bridges connect participants in conference calls. Different types of conference bridges have been used depending in part upon the type of network and how voice is carried over the network to the conference bridge. One type of conference bridge is described in U.S. Pat. No. 5,436,896 (see the entire patent). This conference bridge 10 operates in an environment where voice signals are digitally encoded in a 64Kbps data stream (FIG. 1, col. 1, Ins. 21-26). Conference bridge 10 has a plurality of inputs 12 and outputs 14. Inputs 12 are connected through respective speech detectors 16 and switches 18 to a common summing amplifier 20. Speech detector 16 detects speech by sampling an input data stream and determining the amount of energy present over time. (col. 1, Ins. 36-39). Each speech detector 16 controls a switch 18. When no speech is present switch 18 is held open to reduce noise. During a conference call, inputs 12 of all participants who are speaking are coupled through summing amplifier 20 to each of the outputs 14. Subtracters 24 subtract each participant's own voice data stream. A number of participants 1-n then can speak and hear each other in the connections made through conference bridge 10. See, '896 patent, col. 1, In, 12- col. 2, In. 16. Digitized voice is now also being carried in packets over packet-switched networks. The '896 patent describes one example of asynchronous mode transfer (ATM) packets (also called cells). To support a conference call in this networking environment, conference bridge 10 converts input ATM cells to network packets. Digitized voice is extracted from the packets and processed in conference bridge 12 as described above. The summed output digitized voices is re-converted from network packets back to ATM cells prior to being sent to participants 1-n. See, '896 patent, col. 2, In. 17-col. 2, In. 36. The '896 patent also describes a conference bridge 238 shown in FIGs. 2 and 3 which processes ATM cells without converting and re-converting the ATM cells to network packets as in conference 10. Conference bridge 238 has inputs 302-306, one from each of the participants, and outputs 308-312, one to each of the participants. Speech detectors 314-318 analyze input data aggregated in sample and hold buffers 322-326. Speech detectors 314-318 report the detected speech an/or volume of detected speech to controller 320. See, 4896 patent, col. 4, Ins. 26-39. Controller 320 is coupled to a selector 328, gain control 329 and replicator 330. Controller 320 determines which of the participants is speaking based on the outputs of speech detectors 314-318. When one speaker (such as participant 1) is talking, controller 320 sets selector 328 to read data from buffer 322. The data moves through automatic gain control 329 to replicator 330. Replicator replicates the data in the ATM cell selected by selector 328 for all participants except the speaker. See, '896 patent, col. 4, In. 40-col. 5, In. 5. When two or more speakers are speaking, the loudest speaker is selected in a given selection period. The next loudest speaker is then selected in a subsequent selection period. The appearance of simultaneous speech is kept up by scanning speech detectors 314-318 and reconfiguring selector 328 at appropriate interval such as six milliseconds. See, '896 patent, col. 5, Ins. 6-65. Another type of conference bridge is described in U.S. Pat. No. 5,983,192 (see the entire patent). In one embodiment, a conference bridge 12 receives compressed audio packets through a real-time transport protocol (RTP/RTCP). See, '192 patent, col. 3, In. 66-col. 4, In. 40. Conference bridge 12 includes audio processors 14a-14d. Exemplary audio processor 14c associated with a site C (i.e., a participant C) includes a switch 22 and selector 26. Selector 26 includes a speech detector which determines which of other sites A, B, or D has the highest likelihood of speech. See, '192 patent, col. 4, Ins. 40-67. Alternatives include selecting more than one site and using an acoustic energy detector. See, ' 192 patent, col. 5, Ins. 1-7. In another embodiment described in the' 192 patent, the selector 26/switches 22 output a plurality of loudest speakers in separate streams to local mixing end-point sites. The loudest streams are sent to multiple sites. See, '192 patent, col. 5, Ins. 8-67. Configurations of mixer/encoders are also described to handle multiple speakers at the same time, referred to as "double-talk" and "triple-talk." See, '192 patent, col. 7, In. 20-col. 9, In. 29. Voice-over-the-Intemet (VOIP) systems continue to require an improved conference bridge. For example, a Softswitch VOIP architecture may use one or more media servers having a media gateway control protocol such as MGCP (RFC 2705). See, D. Collins, Carrier Grade Voice over IP, Mc-Graw Hill: United States, Copyright 2001, pp. 234-244, the entire book of which is incorporated in its entirety herein by reference. Such media servers are often used to process audio streams in VOIP calls. These media servers are often endpoints where audio streams are mixed in a conference call. These endpoints are also referred to as "conference bridge access points" since the media server is an endpoint where media streams from multiple callers are mixed and provided again to some or all of the callers. See, D. Collins, p. 242. As the popularity and demand for IP telephony and VOIP calls increases, media servers are expected to handle conference call processing with carrier grade quality. Conference bridges in a media server need to be able to scale to handle different numbers of participants. Audio in packet streams, such as RTP/RTCP packets, needs to be processed in real-time efficiently. BRIEF SUMMARY OF THE INVENTION The present invention provides a method and system for providing media services in Voice over IP telephony. In one embodiment, a switch is coupled between multiple audio sources and a network interface controller. The switch can be a packet switch or a cell switch. Internal and/or external audio sources generate audio streams of packets. Any type of packet can be used. In one embodiment, an internal packet includes a packet header and a payload. In one embodiment, the packet header has information that identifies active speakers whose audio is mixed. The payload carries the digitized mixed audio. According to a feature of the present invention, one fully mixed audio stream of packets and a number of partially mixed audio streams of packets are generated by an audio source (e.g. a DSP). The fully mixed audio stream includes the audio content of a group of identified active speakers. Packet header information identifies each of the active speakers in the fully mixed stream. In one example, the audio source inserts conference identification numbers (CIDs) associated with the respective active speakers into header fields in the packets. The audio source inserts mixed digital audio from the active speakers into the payload of the packets. The mixed digital audio corresponds to speech or other type of audio that is input by the active speakers in the conference call. Each of the partially mixed audio streams includes the audio content of the group of identified active speakers minus the audio content of a respective recipient active speaker. The recipient active speaker is the active speaker within the group of active speakers towards which a partially mixed audio stream is directed. The audio source inserts into the packet payloads the digital audio from the group of identified active speakers minus the audio content of the recipient active speaker. In this way, the recipient active speaker will not receive audio corresponding to their own speech or audio input. Packet header information identifies the active speakers whose audio content is included in the respective partially mixed audio stream. In one example, the audio source inserts one or more conference identification numbers (CIDs) into TAS and IAS header fields of packets. The TAS (Total Active Speakers) field lists CIDs of all of the current active speaker calls in the conference call. The IAS field (Included Active Speakers) lists CIDs of the active speakers whose audio content is in the respective partially mixed stream. In one embodiment, the audio source (or "mixer" since it is mixing audio) dynamically generates the appropriate fully mixed and partially mixed audio streams of packets having CDD information and mixed audio during the conference call. The audio source retrieves the appropriate CID information of conference call participants from a relatively static look-up table generated and stored at the initiation of the conference call. For example, in a conference call where there are 64 participants of which 3 are identified as active speakers (1-3), then one fully mixed audio stream will contain audio from all 3 active speakers. This fully mixed stream is eventually sent to each of the 61 passive participants. Three partially mixed audio streams are also generated. A first partially mixed stream 1 contains audio from speakers 2-3 but not speaker 1. A second partially mixed stream 2 contains audio from speakers 1-3 but not speaker 2. A third partially mixed stream 3 contains audio from speakers 1 -2 but not speaker 3. The first through third partially mixed audio streams are eventually sent to speakers 1-3 respectively. In this way only four mixed audio streams need be generated by the audio source. The fully mixed audio stream and a number of partially mixed audio streams are sent from the audio sources (e.g. DSPs) to a packet switch. A cell layer can also be used. The packet switch multicasts each fully mixed and partially mixed audio stream to a network interface controller (NIC). The NIC then processes each packet to determine whether to forward the packet for the fully mixed or partially mixed audio stream to a participant. This determination can be made in real-time based on a look-up table at the NIC and the packet header information in the multicasted audio streams. In one embodiment, during initialization of the conference call, each participant in the call is assigned a CID. A switched virtual circuit (SVC) is also associated with the conference call participant. A look-up table is generated and stored which includes entries for the conference call participants. Each entry includes the network address information (e.g, IP, UDP address information) and CID of a respective conference call participant. Look-up tables can be stored for access by both the NIC processing packets and the audio source(s) mixing audio during a conference call. The packet switch multicasts each fully mixed and partially mixed audio stream on all of the SVCs assigned to the conference call to the NIC. The NIC processes each packet arriving on the SVCs and in particular examines packet headers to determine whether to discard or forward the packet for the fully mixed or partially mixed audio stream to a participant. One advantage of the present invention is that this packet processing determination can be performed quickly and in real-time during a conference call based on packet header information and CID information obtained from the lookup table. In one embodiment, the network packet which is sent includes the participant's network address information (IP/UDP) obtained from the look-up table, RTP packet header information (timestamp/sequence information), and audio data. In sum, advantages of the present invention include providing conference bridge processing using fewer resources with less bandwidth and processing than is typically required in mixing devices in other conference bridges. A conference bridge system and method of the present invention multicasts in a way that relieves the mixing device of the work of replication. For a conference call with N participants and c active speakers, an audio source only needs to generate c + 1 mixed audio streams (one fully mixed audio stream and c partially mixed audio streams). Work is distributed to the a multicaster in a switch which performs replication and multicasts the mixed audio streams. A further advantage is that a conference bridge according to the present invention can scale to accommodate large numbers of participants. For example, if N = 1000 participants, and c = 3 active speakers then an audio source only need generate c + 1 = 4 mixed audio streams. Packets in multicasted audio streams are processed at a NIC in real-time to determine the appropriate packets for output to a participant in the conference call. In one example, internal egress packets having a header and payload are used in the conference bridge to further reduce processing work at an audio source mixing the audio for the conference call. In addition, as the use of audio networking increases and the number of users and applications rise, there is an increasing need for multiple audio streams even in a given telephone call. The inventors recognized that multiple audio streams need to be switched dynamically without introducing RTP errors in calls placed in an audio networking environment such as a voice over IP network. Such RTP errors can cause unwanted noises such as clicks, pops, etc. The present invention 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. For established VOIP calls, the present invention can noiselessly switch audio from one audio source to another. This switching system is dynamic and can scale to handle many calls. In embodiments of the present invention, a switch is used to direct audio data from multiple audio sources to a network interface controller. The switch can be a cell switch or a packet switch. The audio sources can be internal audio sources and/or external audio sources. The network interface controller (NIC) can be any interface with an IP network and includes one or more packet processors. An egress audio controller controls the operation of internal audio sources, the switch and the network interface controller to carry out noiseless switching according to the present invention. In one feature of the invention, priority information is used by a network interface controller to determine which audio stream from an internal or external audio source is transmitted in an established VOIP telephone call. Consider the case of two internal audio sources. The audio sources generate respective audio streams of internal egress packets for one destination egress audio channel. In one embodiment, each internal egress packet includes a payload carrying audio and control header information. The control header information has priority information. This priority information is then used by a network interface controller to determine which audio stream is transmitted because only one RTP stream can be output at a given time for each VOIP call. In one feature of the invention, the internal egress packets are smaller than IP packets and consist of payload and control header information only. In this way, processing work required to create complete IP packets need not be carried out by internal audio sources such as DSPs but is distributed to the packet processors in the network interface controller. According to further feature, a cell switch is used which is a fully meshed cell switch such as an ATM cell switch that has plenty of available bandwidth. The internal egress packets for the different audio streams are converted to cells. The cell switch combines merged cells from different sources and delivers them across a switched virtual circuit (SVC) to a NIC. The SVC is associated with one egress output audio channel serving an established telephone call. In one embodiment, an egress audio controller is used to control noiseless switching of audio in VOIP telephone calls. This noiseless switching according to the present invention is also referred to herein as a "noiseless switch over." In one embodiment, noiseless switch over of additional audio is carried out for calls in which this service is available. In this way, an extra charge may be made for providing a noiseless switch over service. In other embodiments, noiseless switch over is performed for any call. Certain call events which involve additional audio trigger the noiseless switch over. This noiseless switch over is carried out using the noiseless switching system and method of the present invention. Examples of call events include but are not limited to the following conditions: an emergency condition, a call signaling condition, a call event based on callee or caller information, or a request for different audio information. The request for audio information can be any audio request such as a request for advertisements, news sports, financial, music or other audio content. Audio sources can generate any type of audio. For example, an audio stream of egress packets can include audio payloads representing voice, music, tones, and/or any other sound. The egress audio controller can be a stand-alone unit or a part of a call control and audio feature manager in an audio processing platform. The present invention can be implemented in a media server, audio processor, router, packet switch, or audio processing platform. Another embodiment involves the switching of audio streams including an audio stream from an external audio source. In this case, a NIC receives IP packets containing the audio stream and converts the IP packets to internal egress packets. At this point, the internal egress packets are processed as if they were generated by an internal audio source. The internal egress packets may include priority information. The internal egress packets may be sent as packets or cells across a SVC through a switch to the NIC. When the external audio stream has a relatively high priority and switch over is to proceed, a packet processor at the NIC generates IP packets with synchronized header information (such as RTP information) and sends the IP packets to a destination device. In one embodiment, a noiseless switch over system according to the invention involves the switching of audio streams only from internal audio sources such as DSPs. In another embodiment, a noiseless switch over system according to the invention involves the switching of audio streams from internal audio sources and external audio sources. In another embodiment, a noiseless switch over system according to the invention involves the switching of audio streams only from external audio sources in which case the switch over system acts a general switch for audio streams and no internal DSPs are required. Further embodiments, features, and advantages of the present inventions, as well as the structure and operation of the various embodiments of the present invention, are described in detail below with reference to the accompanying drawings. BRIEF DESCRIPTION OF THE ACCOMPANYING DRAWINGS The accompanying drawings, which are incorporated herein and form a part of the specification, illustrate the present invention and, together with the description, further serve to explain the principles of the invention and to enable a person skilled in the pertinent art to make and use the invention. In the drawings: FIG. 1 is a diagram of a media server in a voice over the Internet example environment according to the present invention. FIG. 2 is a diagram of an example media server including media services and resources according to the present invention. FIGs. 3 A and 3B are diagrams of an audio processing platform according to an embodiment of the present invention. FIG. 4 is a diagram of a audio processing platform as shown in FIG. 3 according to an example implementation of the present invention. FIG. 5A is a flow diagram showing the establishment of a call and ingress packet processing according to an embodiment of the present invention. FIG. 5B is a flow diagram showing egress packet processing and call completion according to an embodiment of the present invention. FIGs. 6A-6F are diagrams of noiseless switch over systems according to embodiments of the present invention. FIG. 6A is diagram of a noiseless switch over system that carries out cell switching of independent egress audio streams generated by internal audio sources according to an embodiment of the present invention. FIG. 6B is diagram of audio data flow in a noiseless switch over system that carries out cell switching of independent egress audio streams generated by internal audio sources according to an embodiment of the present invention. FIG. 6C is diagram of a noiseless switch over system that carries out cell switching between independent egress audio streams generated by internal and/or external audio sources according to an embodiment of the present invention. FIG. 6D is diagram of audio data flow in a noiseless switch over system that carries out cell switching between independent egress audio streams generated by internal and/or external audio sources according to an embodiment of the present invention. FIG. 6E is diagram of audio data flow in a noiseless switch over system that carries out packet switching between independent egress audio streams generated by internal and/or external audio sources according to an embodiment of the present invention. FIG. 6F is diagram of a noiseless switch over system that carries out switching between independent egress audio streams generated by external audio sources according to an embodiment of the present invention. FIG. 7A is a schematic illustration of an IP packet with RTP information. FIG. 7B is a schematic illustration of an internal packet according to one embodiment of the present invention. FIG. 8 is a flow diagram showing the switching functionality according to one embodiment of the present invention. FIG. 9A, 9B, and 9C are flow diagrams showing the call event processing for audio stream switching according to one embodiment of the present invention. FIG. 10 is a block diagram of a distributed conference bridge according to one embodiment of the present invention. FIG. 11 is an example look-up table used in the distributed conference bridge of FIG. 10. FIG. 12 is a flowchart diagram of the operation of the distributed conference bridge of FIG. 10 in establishing a conference call. FIGs. 13A, 13B, and 13C are flowchart diagrams of the operation of the distributed conference bridge of FIG. 10 in processing a conference call. FIG. 14A is a diagram of an example internal packet generated by an audio source during a conference call according to one embodiment of the present invention. FIG. 14B is a diagram that illustrates example packet content in a fully mixed audio stream and set of partially mixed audio streams according to the present invention. FIG. 15 is a diagram that illustrates example packet content after the packets of FIG. 14 have been multicasted and after they have been processed into IP packets to be sent to appropriate participants in a 64 participant conference call according to the present invention. The present invention will now be described with reference to the accompanying drawings. In the drawings, like reference numbers indicate identical or functionally similar elements. Additionally, the left-most digit(s) of a reference number identifies the drawing in which the reference number first appears. DETAILED DESCRIPTION OF THE INVENTION I. Overview and Discussion The present invention provides a method and system for distributed conference bridge processing in Voice over IP telephony. Work is distributed away from a mixing device such as a DSP. In particular, a distributed conference bridge according to the present invention uses internal multicasting and packet processing at a network interface to reduce work at an audio mixing device. A conference call agent is used to establish and end a conference call. An audio source such as a DSP mixes audio of active conference call participants. Only one fully mixed audio stream and a set of partially mixed audio streams need to be generated. A switch is coupled between the audio source mixing audio content and a network interface controller. The switch includes a multi-caster. The multi-caster replicates packets in the one fully mixed audio stream and a set of partially mixed audio streams and multi-casts the replicated packets to links (such as SVCs) associated with each call participant. A network interface controller processes each packet to determine whether to discard or forward the packet for the fully mixed or partially mixed audio stream to a participant. This determination can be made in real-time based on a look-up table at the NIC and the packet header information in the multicasted audio streams. In one embodiment, a conference bridge according to the present invention is implemented in a media server. According to embodiments of the present invention, the media server can include a call control and audio feature manager for managing the operations of the conference bridge. The present invention is described in terms of an example voice over the Internet environment. Description in these terms is provided for convenience only. It is not intended that the invention be limited to application in these example environments. In fact, after reading the following description, it will become apparent to a person skilled in the relevant art how to implement the invention in alternative environments known now or developed in the future. II. Terminology To more clearly delineate the present invention, an effort is made throughout the specification to adhere to the following term definitions as consistently as possible. The term noiseless according to the present invention refers to switching between independent audio streams where packet sequence information is preserved. The term synchronized header information refers to packets having headers where packet sequence information is preserved. Packet sequence information can include but is not limited to valid RTP information. The term digital signal processor (DSP) includes but is not limited to a device used to code or decode digitized voice samples according to a program or application service. The term digitized voice or voice includes but is not limited to audio byte samples produced in a pulse code modulation (PCM) architecture by a standard telephone circuit compressor/decompressor (CODEC). The term packet processor refers to any type of packet processor that creates packets for a packet-switched network. In one example, a packet processor is a specialized microprocessor designed to examine and modify Ethernet packets according to a program or application service. The term packetized voice refers to digitized voice samples carried within a packet. The term real time protocol (RTP) stream of audio refers to the sequence of RTP packets associated with one channel of packetized voice. The term switched virtual circuit (SVC) refers to a temporary virtual circuit that is set up and used only as long as data is being transmitted. Once the communication between the two hosts is complete, the SVC disappears. In contrast, a permanent virtual circuit (PVC) remains available at all times. HI. Audio Networking Environment The present invention can be used in any audio networking environment. Such audio networking environments can include but are not limited to a wide area and/or local area network environment. In example embodiments, the present invention is incorporated within an audio networking environment as a stand-alone unit or as part of a media server, packet router, packet switch or other network component. For brevity, the present invention is described with respect to embodiments incorporated in a media server. Media servers deliver audio on network links over one or more circuit- switched and/or packet-switched networks to local or remote clients. A client can be any type of device that handles audio including but not limited to a telephone, cellular phone, personal computer, personal data assistant (PDA), set-top box, console, or audio player. FIG. 1 is a diagram of a media server 140 in an voice over the Internet example environment according to the present invention. This example includes a telephone client 105, public-switched telephone network (PSTN) 110, Softswitch 120, gateway 130, media server 140, packet-switched network(s) 150, and computer client 155. Telephone client 105 is any type of phone (wired or wireless) that can send and receive audio over PSTN 110. PSTN 110 is any type of circuit-switched network(s). Computer client 155 can be a personal computer. Telephone client 105 is coupled through a public-switched telephone network (PSTN) 110, gateway 130 and network 150 to media server 140. In this example, call signaling and control is separated from the media paths or links that carry audio. Softswitch 120 is provided between PSTN 110 and media server 140. Softswitch 120 supports call signaling and control to establish and remove voice calls between telephone client 105 and media server 140. In one example, Softswitch 120 follows the Session Initiation Protocol (SIP). Gateway 130 is responsible for converting audio passing to and from PSTN 110 and network 150. This can include a variety of well-known functions such as translating a circuit- switched telephone number to an Internet Protocol (IP) address and vice versa. Computer client 155 is coupled over network 150 to media server 140. A media gateway controller (not shown) can also use SIP to support call signaling and control to establish and breakdown links such as voice calls between computer client 155 and media server 140. An application server(not shown) can also be coupled to media server 140 to support VOL? services and applications. The present invention is described in terms of these example environments. Description in these terms is provided for convenience only. It is not intended that the invention be limited to application in these example environments involving a media server, router, switch, network component, or stand-alone unit within a network. In fact, after reading the following description, it will become apparent to a person skilled in the relevant art how to implement the invention in alternative environments known now or developed in the future. IV. Media Server, Services and Resources FIG. 2 is a diagram of an example media platform 200 according to one embodiment the present invention. Platform 200 provides scalable VOIP telephony. Media platform 200 includes a media server 202 coupled to resource(s) 210, media service(s) 212, and interface(s) 208. Media server 202 includes one or more applications 210, a resource manager 220 and audio processing platform 230. Media server 202 provides resources 210 and services 212. Resources 210 include, but are not limited to modules 21 la-f, as shown in FIG 2. Resource modules 21 la-f include conventional resources such as play announcements/collect digits IVR resources 211a, tone/digit voice scanning resource 21 lb, transcoding resource 21 lc, audio record/play resource 21 Id, text- to-speech resource 21 le, and speech recognition resource 21 If. Media services 212 include, but are not limited to, modules 213a-e, as shown in FIG. 2. Media services modules 213a-e include conventional services such as telebrowsing 213a, voice mail service 213b, conference bridge service 213c, video streaming 213d, and a VOIP gateway 213e. Media server 202 includes an application central processing unit (CPU) 210, a resource manager CPU 220, and an audio processing platform 230. Application CPU 210 is any processor that supports and executes program interfaces for applications and applets. Application CPU 210 enables platform 200 to provide one or more of the media services 212. Resource manager CPU 220 is any processor that controls connectivity between resources 210 and the application CPU 210 and/or audio processing platform 230. Audio processing platform 230 provides communications connectivity with one or more of the network interfaces 208. Media platform 200 through audio processing platform 230 receives and transmits information via network interface 208. Interface 208 can include, but it not limited to, Asynchronous Transfer Mode (ATM) 209a, local area network (LAN) Ethernet 209b, digital subscriber line (DSL) 209c, cable modem 209d, and channelized T1-T3 lines 209e. V. Audio Processing Platform with a Packet/Cell Switch for Noiseless Switching of Independent Audio Streams In one embodiment of the present invention, audio processing platform 230 includes a dynamic fully-meshed cell switch 304 and other components for the reception and processing of packets, such as Internet Protocol (IP) packets. Platform 230 is shown in FIG. 3A with regard to audio processing including noiseless switching according to the present invention. As illustrated, audio processing platform 230 includes a call control and audio feature manager 302, cell switch 304 (also referred to as a packet/cell switch to indicate cell switch 304 can be a cell switch or packet switch), network connections 305, network interface controller 306, and audio channel processors 308. Network interface controller 306 further includes packet processors 307. Call control and audio feature manager 302 is coupled to cell switch 304, network interface controller 306, and audio channels processors 308. In one configuration, call control and audio feature manager 302 is connected directly to the network interface controller 306. Network interface controller 306 then controls packet processor 307 operation based on the control commands sent by call control and audio feature manager 302. In one embodiment, call control and audio feature manager 302 controls cell switch 304, network interface controller 306 (including packet processors 307), and audio channel processors 308 to provide noiseless switching of independent audio streams according to the present invention. This noiseless switching is described further below with respect to FIGs. 6-9. An embodiment of the call control and audio feature manager 302 according to the present invention is described further below with respect to FIG. 3B. Network connections 305 are coupled to packet processors 307. Packet processors 307 are also coupled to cell switch 304. Cell switch 304 is coupled in tum to audio channel processors 308. In one embodiment, audio channel processors 308 include four channels capable of handling four calls, i.e., there are four audio processing sections. In alternative embodiments, there are more or less audio channel processors 308. Data packets, such as IP packets, that include payloads having audio data arrive at network connections 305. In one embodiment, packet processors 307 comprise one or more or eight 100Base-TX full-duplex Ethernet links capable of high speed network traffic in the realm of 300,000 packets per second per link. In another embodiment, packet processors 307 are capable of 1,000 G.711 voice ports per link and/or 8,000 G.711 voice channels per system. In additional embodiments, packet processors 307 recognize the IP headers of packets and handle all RTP routing decisions with a minimum of packet delay or jitter. In one embodiment of the present invention, packet/cell switch 304 is a non-blocking switch with 2.5Gbps of total bandwidth. In another embodiment, the packet/cell switch 204 has 5Gbps of total bandwidth. In one embodiment, the audio channel processors 308 comprise any audio source, such as digital signal processors, as described in further detail with regards to FIG. 4. The audio channel processors 308 can perform audio related services including one or more of the services 211a-f. VI. Example Audio Processing Platform Implementation FIG. 4 shows one example implementation which is illustrative and not intended to limit the present invention. As shown in FIG. 4, audio processing platform 230 can be a shelf controller card (SCC). System 400 embodies one such SCC. System 400 includes cell switch 304, call control and audio feature manager 302, a network interface controller 306, interface circuitry 410, and audio channel processors 308a-d. More specifically,, system 400 receives packets at network connections 424 and 426. Network connections 424 and 426 are coupled to network interface controller 306. Network interface controller 306 includes packet processors 307a-b. Packet processors 307a-b comprise controllers 420, 422, forwarding tables 412,416, and forwarding processor (EPIF) 414,418. As shown in FIG. 4, packet processor 307a is coupled to network connection 424. Network connection 424 is coupled to controller 420. Controller 420 is coupled to both forwarding table 412 andEPIF414. Packet processor 307b is coupled to network connection 426. Network connection 426 is coupled to controller 422. Controller 422 is coupled to both forwarding table 416 and EPIF 418. In one embodiment, packet processors 307 can be implemented on one or more LAN daughtercard modules. In another embodiment, each network connection 424 and 426 can be a 100Base-TX or 1000Base-T link. The IP packets received by the packet processors 307 are processed into internal packets. When a cell layer is used, the internal packets are then converted to cells (such as ATM cells' by a conventional segmentation and reassembly (SAR) module). The cells are forwarded by packet processors 307 to cell switch 304. The packet processors 307 are coupled to the cell switch 304 via cell buses 428,430,432,434. Cell switch 304 forwards the cells to interface circuitry 410 via cell buses 454,456,458,460. Cell switch 304 analyzes each of the cells and forwards each of the cells to the proper cell bus of cell buses 454, 456, 458, 460 based on an audio channel for which that cell is destined. Cell switch 304 is a dynamic, fully-meshed switch. In one embodiment, interface circuitry 410 is a backplane connector. The resources and services available for the processing and switching of the packets and cells in system 400 are provided by call control and audio feature manager 304. Call control and audio feature manager 302 is coupled to cell switch 402 via a processor interface (PDF) 436, a SAR, and a local bus 437. Local bus 437 is further coupled to a buffer 438. Buffer 438 stores and queues instructions between the call control and'audio feature manager 302 and the cell switch 304. Call control and audio feature manager 302 is also coupled to a memory module 442 and a configuration module 440 via bus connection 444. In one embodiment, configuration module 440 provides control logic for the boot-up, initial diagnostic, and operational parameters of call control and audio feature manager 302. In one embodiment, memory module 442 comprises dual in-line memory modules (DIMMs) for random access memory (RAM) operations of call control and audio feature manager 302. Call control and audio feature manager 302 is further coupled to interface circuitry410. A network conduit 408 couples resource manager CPU 220 and/or application CPU 210 to the interface circuitry 410. In one embodiment, call control and audio feature manager 302 monitors the status of the interface circuitry 410 and additional components coupled to the interface circuitry 410. In another embodiment, call control and audio feature manager 302 controls the operations of the components coupled to the interface circuitry 410 in order to provide the resources 210 and services 212 of platform 200. A console port 470 is also coupled to call control and audio feature manager 302. Console port 470 provides direct access to the operations of call control and audio feature manager 302. For example, one could administer the operations, re-boot the media processor, or otherwise affect the performance of call control and audio feature manager 302 and thus the system 400 using the console port 470. Reference clock 468 is coupled to interface circuitry 410 and other components of the system 400 to provide consistent means of time-stamping the packets, cells and instructions of the system 400. Interface circuitry 410 is coupled to each of audio channel processors 308a-308d. Each of the processors 308 comprise a PIF 476, a group 478 of one or more card processors (also referred to as "bank" processors), and a group 480 of one or more digital signal processors (DSP) and SDRAM buffers. In one embodiment, there are four card processors in: group 478 and 32 DSPs in group 480. In such an embodiment, each card processor of group 478 would access and operate with eight DSPs of group 480. VII. Call Control and Audio Feature Manager FIG. 3B is a block diagram of call control and audio feature manager 302 according to one embodiment of the present invention. Call control and audio feature manager 302 is illustrated functionally as processor 302. Processor 302 comprises a call signaling manager 352, system manager 354, connection manager 356, and feature controller 358. Call signaling manager 352 manages call signaling operation such as call establishment and removal, interface with a Softswitch, and handling signaling protocols like SIP. System manager 354 performs bootstrap and diagnostic operations on the components of system 230. System manager 354 further monitors the system 230 and controls various hot-swapping and redundant operation. Connection manager 356 manages EPIF forwarding tables, such as tables 412 and 416, and provides the routing protocols (such as Routing Information Protocol (RIP), Open Shortest Path First (OSPF), and the like). Further, the connection manager 356 establishes internal ATM permanent virtual circuits (PVC) and/or SVC. In one embodiment, the connection manager 356 establishes bi-directional connections between the network connections, such as network connections 424 and 426, and the DSP channels, such as DSPs 480a-d, so that data flows can be sources or processed by a DSP or other type of channel processor. In another embodiment, connection manager 356 abstracts the details of the EPIF and ATM hardware. Call signaling manager 352 and the resource manager CPU 220 can access these details so that their operations are based on the proper service set and performance parameters. Feature controller 358 provides communication interfaces and protocols such as, H.323, and MGCP (Media Gateway Control Protocol). In one embodiment, card processors 478a-d function as controllers with local managers for the handling of instructions from the call control and audio feature manager 302 and any of its modules: call signaling manager 352, system manager 354, connection manager 356, and feature controller 358. Card processors 478a-d then manage the DSP banks, network interfaces and media streams, such as audio streams. In one embodiment, the DSPs 480a-d provide the resources 210 and services 212 of platform 200. In one embodiment, call control and audio feature manager 302 of the present invention exercises control over the EPIFof the present invention through the use of applets. In such an embodiment, the commands for configuring parameters (such as port MAC address, port IP address, and the like), search table management, statistics uploading, and the like, are indirectly issued through applets. The EPIF provides a search engine to handle the functionality related to creating, deleting and searching entries. Since the platform 200 operates on the source and destination of packets, the EPIF provides search functionality of sources and destinations. The sources and destinations of packets are stored in search tables for incoming (ingress) and outgoing (egress) addresses. The EPIF can also manage RTP header information and evaluating relative priorities of egress audio streams to be transmitted as described in further detail below. VIII. Audio Processing Platform Operation The operation of audio processing platform 230 is illustrated in the flow diagrams of FIGs. 5A and 5B. FIG. 5A is a flow diagram showing the establishment of a call and ingress packet processing according to an embodiment of the present invention. FIG. 5B is a flow diagram showing egress packet processing and call completion according to an embodiment of the present invention. A. Ingress Audio Streams In FIG. 5A, the process for an ingress (also called inbound) audio stream starts at step 502 and immediately proceeds to step 504. In step 504, call control and audio feature manager 302 establishes a call with a client communicating via the network connections 305. In one embodiment, call control and audio feature manager 302 negotiates and authorizes access to the client. Once client access is authorized, call control and audio feature manager 302 provides IP and UDP address information for the call to the client. Once the call is established, the process immediately proceeds to step 506. In step 506, packet processors 307 receive IP packets carrying audio via the network connections 305. Any type of packet can be used including but not limited to IP packets, such as Appletalk, IPX, or other type of Ethernet packets. Once a packet is received, the process proceeds to step 508. In step 508, packet processors 307 check IP and UDP header address in search table to find associated SVC, and then convert the VOIP packets into interna] packets. Such internal packets for example can be made up of a payload and control header as described further below with respect to FIG. 7B. Packet processors 307 then construct packets using at least some of the data and routing information and assign a switched virtual circuit (SVC). The SVC is associated with one of the audio channel processors 308, and in particular with one of respective DSP that will process the audio payload. When a cell layer is used, internal packets are further converted or merged into cells, such as ATM cells. In this way, audio payloads in the internal packets are converted to audio payloads in a stream of one or more ATM cells. A conventional segmentation and reassembly (SAR) module can be used to convert internal packets to ATM cells. Once the packets are converted into the cells, the process proceeds to step 510. In step 510, cell switch 304 switches the cells to the proper audio channel of the audio channel processors 308 based on the SVC. The process proceeds to step 512. In step 512, audio channel processors 308 convert the cells into packets. Audio payloads in the arriving ATM cells for each channel are converted to audio payloads in a stream of one or more packets. A conventional SAR module can be used to convert ATM to packets. Packets can be internal egress packets or IP packets with audio payloads. Once the cells are converted into the internal packets, the process proceeds to step 514. In step 514, audio channel processors 308 process the audio data of the packets in the respective audio channels. In one embodiment, the audio channels are related to one or more of the media services 213a-e. For example, these media services can be telebrowsing, voice mail, conference bridging (also called conference calling), video streaming, VOIP gateway services, telephony, or any other media service for audio content. B. Egress Audio Streams In FIG. 5B, the process for an egress (also called outbound) audio stream starts at step 522 and immediately proceeds to step 524. In step 524, call control and audio feature manager 302 identifies an audio source for noiseless switch over. This audio source can be associated with an established call or other media service. Once the audio source is identified, the process immediately proceeds to step 526. In step 526, an audio source creates packets. In one embodiment, a DSP in audio channel processor 308 is an audio source. Audio data can be stored in a SDRAM associated with the DSP. This audio data is then packetized by a DSP into packets. Any type of packet can be used including but not limited to internal packets or IP packets, such as Ethernet packets. In one preferred embodiment, the packets are internal egress packets generated as described with respect to FIG. 7B. In step 528, an audio channel processor 308 converts the packets into cells, such as ATM cells. Audio payloads in the packets are converted to audio payloads in a stream of one or more ATM cells. In brief, the packets are parsed and the data and routing information analyzed. Audio channel processor 308 then construct cells using at least some of the data and routing information and assigns a switched virtual circuit (SVC). A conventional SAR module can be used to convert packets to ATM cells. The SVC is associated with one of the audio channel processors 308, and in particular with a circuit connecting the respective DSP of the audio source and a destination port 305 of NIC 306. Once the packets are converted into the cells, the process proceeds to step 530. In step 530, cell switch 304 switches the cells of an audio channel of the audio channel processors 308 to a destination network connection 305 based on the SVC. The process proceeds to step 532. In step 532, packet processors 307 convert the cells into IP packets. Audio payloads in the arriving ATM cells for each channel are converted to audio payloads in a stream of one or more internal packets. A conventional SAR module can be used to convert ATM to internal packets. Any type of packet can be used including but not limited to IP packets, such as Ethernet packets. Once the cells are converted into the packets, the process proceeds to step 534. In step 534, each packet processor 307 further adds RTP, IP, and UDP header information. A search table is checked to find IP and UDP header address information associated with the SVC. IP packets are then sent carrying audio via the network connections 305 over a network to a destination device (phone, computer, palm device, PDA, etc.). Packet processors 307 process the audio data of the packets in the respective audio channels. In one embodiment, the audio channels are related to one or more of the media services 213a-e. For example, these media services can be telebrowsing, voice mail, conference bridging (also called conference calling), video streaming, VOIP gateway services, telephony, or any other media service for audio content. IX. Noiseless Switching of Egress Audio Streams According to the one aspect of the present invention, audio processing platform 230 noiselessly switches between independent egress audio streams. Audio processing platform 230 is illustrative. The present invention as it relates to noiseless switching of egress audio stream can be used in any media server, router, switch, or audio processor and is not intended to be limited to audio processing platform 230. A. Cell Switch - Internal Audio Sources FIG. 6A is diagram of a noiseless switch over system that carries out cell switching of independent egress audio streams generated by internal audio sources according to an embodiment of the present invention. FIG. 6A shows an embodiment of a system 600A for egress audio stream switching from internal audio sources. System 600A includes components of audio processing platform 230 configured for an egress audio stream switching mode of operation. In particular, as shown in FIG. 6A, system 600A includes call control and audio feature controller 302 coupled to a number n of internal audio sources 604n, cell switch 304, and network interface controller 306. Internal audio sources 604a- 604n can be two or more audio sources. Any type of audio source can be used including but not limited to DSPs. In one example, DSPs 480 can be audio sources. To generate audio, audio sources 604 can either create audio internally and/or convert audio received from external sources. Call control and audio feature controller 302 further includes an egress audio controller 610. Egress audio controller 610 is control logic that issues control signals to audio sources 604n, cell switch 304, and/or network interface controller 306 to carry out noiseless switching between independent egress audio streams according to the present invention. The control logic can implemented in software, firmware, microcode, hardware or any combination thereof. A cell layer including SARs 630, 632,634 is also provided. SARs 630, 632 are coupled between cell switch 304 and each audio source 604a-n. SAR 634 is coupled between cell switch 304 and NIC 306. In one embodiment, independent egress audio streams involve streams of IP packets with RTP information and internal egress packets. Accordingly, it is helpful to first describe IP packets and internal egress packets (FIGs. 7A-7B). Next, system 600A and its operation is described in detail with respect to independent egress audio streams (FIGs. 8-9). B. Packets In one embodiment, the present invention uses two types of packets: (1) IP packets with RTP information and (2) internal egress packets. Both of these types of packets are shown and described with respect to examples in FIGs. 7A and 7B. IP packets 700A are sent and received over a external packet-switched network by packet processors 307 in NIC 306. Internal egress packets 700B are generated by audio sources (e.g. DSPs) 604a-604n. 1. IP Packets with RTP Information A standard Internet Protocol (IP) packet 700A is shown in FIG. 7A. IP packet 700A is shown with various components: media access control (MAC) field 704, IP field 706, user datagram protocol (UDP) field 708, RTP field 710, payload 712 containing digital data, and cyclic redundancy check (CRC) field 714. Real-Time Transport Protocol (RTP) is a standardized protocol for carrying periodic data, such as digitized audio, from a source device to a destination device. A companion protocol, Real-Time Control Protocol (RTCP), can also be used with RTP to provide information on the quality of a session. More specifically, the MAC 704 and IP 706 fields contain addressing information to allow each packet to traverse an IP network interconnecting two devices (origin and destination). UDP field 708 contains a 2-byte port number that identifies a RTP/audio stream channel number so that it can be internally routed to the audio processor destination when received from the network interface. In one embodiment of the present invention, the audio processor is a DSP, as described herein. RTP field 710 contains a packet sequence number and timestamp. Payload 712 contains the digitized audio byte samples and can be decoded by the endpoint audio processors. Any payload type and encoding scheme for audio and/or video types of media compatible with RTP can be used as would be apparent to a person skilled in the art given this description. CRC field 714 provides a way to verify the integrity of the entire packet. See, the description of RTP packets and payload types described by D. Collins, Carrier Grade Voice over IP, pp. 52-72 (the text of the entire book of which is incorporated herein by reference). 2. Internal Egress Packets FIG. 7B illustrates an example internal egress packet of the present invention in greater detail. Packet 700B includes a control (CTRL) header 720 and a payload 722. The advantage of internal egress packet 700B is it is simpler to create and smaller in size than IP packet 700A. This reduces the burden and work required of audio sources and other components handling the internal egress packets. In one embodiment, audio sources 604a-604n are DSPs. Each DSP adds a CTRL header 720 in front of a payload 722 that it creates in for a respective audio stream. CTRL 720 is then used to relay control information downstream. This control information for example can be priority information associated with a particular egress audio stream. Packet 700B is converted to one or more cells, such as ATM cells, and sent internally over cell switch 304 to a packet processor 307 in network interface controller 306. After the cells are converted to internal egress packets, packet processor 307 decodes and removes internal header CTRL 720. The rest of the IP packet information is added before the payload 722 is transmitted as an IP packet 700A onto an IP network. This achieves an advantage as processing work at the DSPs is reduced. DSPs only have to add a relatively short control header to payloads. The remaining processing work of adding information to create valid IP packets with RTP header information can be distributed to packet processors) 307. C. Priority Levels Network interface controller (NIC) 306 processes all internal egress packets, as well as all egress IP packets destined for the external network. Thus, NIC 306 can make final forwarding decisions about each packet sent to it based on the content of each packet. In some embodiments, NIC 306 manages the forwarding of egress IP packets based on priority information. This can include switching over to an audio stream of egress IP packets with a higher priority and buffering or not forwarding another audio stream of egress IP packets with a lower priority. In one embodiment, internal audio sources 604a-604n determine priority levels. Alternatively, NIC 306 can determine a priority for audio received from an external source at NIC 306. Any number of priority levels can be used. The priority levels distinguish the relative priority of audio sources and their respective audio streams. Priority levels can be based on any criteria selected by a user including, but not limited to, time of day, identity or group of the caller or callee, or other similar factors relevant to audio processing and media services. Components of the system 600 filter and forward the priority level information within the audio stream. In one embodiment, a resource manager in system 600 can interact with external systems to alter the priority levels of audio streams. For example, an external system can be an operator informing the system to queue a billing notice or advertisement on a call. Thus, the resource manager is capable of barging into audio streams. This noiseless switch over can be triggered by user or automatically based on certain predefined events such as signaling conditions like on-hold condition, emergency event, or timed event. D. Noiseless Fully Meshed Cell Switch System 600A can be thought of as a "free pool" of multiple input (ingress) and output (egress) audio channels because a fully meshed packet/cell switch 304 is used to switch egress audio channels to participate in any given call. Any egress audio channel can be called upon to participate in a telephone call at any time. During both the initial call setup and while the call is in session, any egress audio channel can be switched into and out of the call. The fully meshed switching capability of system 600A of the present invention provides a precise noiseless switching functionality which does not drop or corrupt the IP packets or the cells of the present invention. In addition, a two-stage egress switching technique is used. E. Two-Stage Egress Switching System 600A includes at least two stages of switching. In terms of egress switching, the first stage is cell switch 304. The first stage is cell-based and uses switched virtual circuits (SVCs) to switch audio streams from separate physical sources (audio sources 604a-604n) to a single destination egress network interface controller (NIC 306). Priority information is provided in the CTRL header 720 of cells generated by the audio sources. The second stage is contained within the egress NIC 306 such that it selects which of the audio streams from multiple audio sources (604a-604n) to process and send over a packet network such as an packet-switched IP network. This selection of which audio streams to forward can be performed by NIC 306 is based on the priority information provided in the CTRL headers 720. In this way, a second audio stream with a higher priori ty can be forwarded by NIC 306 on the same channel as a first audio stream. From the perspective of the destination device receiving the audio streams, the insertion of the second audio stream on the channel is received as a noiseless switch between independent audio streams. More specifically, in one embodiment, the egress audio switching can occur in a telephone call. A call is first established using audio source 604a by negotiating with the destination device's MAC, IP, and UDP information, as previously described. First audio source 604a begins generating a first audio stream during the call. The first audio stream is made up of internal egress packets having audio payload and CTRL header 720 information as described with respect to packet format 700B. Internal egress packets egress on the channel established for the call. Any type of audio payload including voice, music, tones, or other audio data can be used. SAR 630 converts the internal packets to cells for transport through cell switch 304 to SAR 634. SAR 634 then converts cells back to internal egress packets prior to delivery to NIC 306. During the flow from the audio source 604a, NIC 306 is decoding and removing the CTRL header 720 and adding the appropriate RTP, UDP, IP, MAC, and CRC fields, as previously described. CTRL header 720 includes the priority field used by NIC 306 to process the packet and send a corresponding RTP packet. NIC 306 evaluates the priority field. Given the relatively high priority field (the first audio source 604a is the only transmitting source), NIC 306 forwards IP packets with synchronized RTP header information which carry the first audio stream over the network to the destination device associated with the call. (Note CTRL header 720 can also include RTP or other synchronized header information which can be used or ignored by NIC 306 if NIC 306 generates and adds RTP header information). When the egress audio controller 610 determines a call event where a noiseless switch over is to occur, a second audio source 604n begins generating a second audio stream. Audio can be generated by audio source 604n directly or by converting audio originally generated by external devices. The second audio stream is made up of internal egress packets having audio payload and CTRL header 720 information as described with respect to packet format 700B. Any type of audio payload including voice, music, or other audio data can be used. Assume the second audio stream is given a higher priority field than the first audio stream. For example, the second audio stream can represent an advertisement, emergency public service message, or other audio data that is desired to have noiselessly inserted into the first channel established with the destination device. The second audio stream's internal egress packets are then converted to cells by SAR 632. Cell switch 304 switches the cells to an SVC destined for the same destination NIC 306 as the first audio stream. SAR 634 converts the cells back to internal packets. NIC 306 now receives the internal packets for the first and second audio streams. NIC 306 evaluates the priority field in each stream. The second audio stream having internal packets with the higher priority are converted to IP packets with synchronized RTP header information and forwarded to the destination device. The first audio stream having internal packets with the lower priority are either stored in a buffer or converted to IP packets with synchronized RTP header information and stored in buffer. NIC 306 can resume forwarding the first audio stream when the second audio stream is completed, after a predetermined time elapses, or when a manual or automatic control signal is received to resume. F. Call Event Triggering Noiseless Switch Over The functionality of the priority field in an embodiment of noiseless switching according to the present invention is now described with regard to FIGs. 8, 9A and 9B. In FIG. 8, a flow diagram of a noiseless switching routine 800 according to one embodiment of the present invention is shown. For brevity, the noiseless switching routine 800 is described with respect system 600. Flow 800 begins at step 802 and proceeds immediately to step 804. In step 804, call control and audio feature manager 302 establishes a call from a first audio source 604a to a destination device. Call control and audio feature manager 302 negotiates with the destination device to determine the MAC, IP and UDP port to use in a first audio stream of IP packets sent over a network. Audio source 604a delivers a first audio stream on one channel for the established call. In one embodiment, a DSP delivers the first audio stream of internal egress packets on one channel to cell switch 304 and then to NIC 306. The process proceeds to step 806. In step 806, egress audio controller 610 sets a priority field for the first audio source. In one embodiment, egress audio controller 610 sets the priority field to a value of one. In another embodiment, the priority field is stored in the CTRL header of the internally routed internal egress packets. The process immediately proceeds to step 808. In step 808, egress audio controller 610 determines the call's status In one embodiment, egress audio controller 610 determines whether or not the call allows or has been configured to allow call events to interact with it. In one embodiment of the present invention, a call can be configured so that only emergency call events will interrupt it. In another embodiment, a call can be configured to receive certain call events based on either the caller(s) or callee(s) (i.e., the one or more of the parties on the call). The process immediately proceeds to step 810. In step 810, egress audio controller 610 monitors for call events. In one embodiment, a call event can be generated within the system 600, such as notifications of time, weather, advertisements, billing ("please insert another coin" or "you have 5 minutes remaining"). In another embodiment, call events can be sent to the system 600, such as requests for news, sporting information, etc. Egress audio controller 610 can monitor both internally and externally for call events. The process proceeds immediately to step 812. In step 812, egress audio controller 610 receives a call event. If not, then egress audio controller 610 continues to monitor as stated in step 810. If so, then the process proceeds immediately to step 814. In step 814, egress audio controller 610 determines the call event and performs the operations necessitated by the call event. The process then proceeds to step 816 where it either ends or returns to step 802. In one embodiment, the process 800 repeats for as long as the call continues. In FIG. 9A-9C, flow diagram 900 of the call event processing for audio stream switching based on priority according to one embodiment of the present invention are shown. In one embodiment, flow 900 shows in more detail the operations performed in step 814 of FIG. 8. Process 900 starts at step 902 and proceeds immediately to step 904. In step 904, egress audio controller 610 reads a call event for an established call. In this operation, a first audio stream from source 604a is already being sent from NIC 306 to a destination device as part of the established call. The process proceeds to step 906. In step 906, egress audio controller 610 determines whether the call event includes a second audio source. If so, then the process proceeds to step 908. If not, then the process proceeds to step 930. In step 908, egress audio controller 610 determines the priority of the second audio source. In one embodiment, egress audio controller 610 issues a command to second audio source 604n that instructs the second audio source to generate a second audio stream of internal egress packets. Priority information for the second audio stream can be automatically generated by the second audio source 604n or generated based on a command from the egress audio controller 610. The process then proceeds to step 910. In step 910, a second audio source 604n begins generating a second audio stream. The second audio stream is made up of internal egress packets having audio payload and CTRL header 720 information as described with respect to packet format 700B. Any type of audio payload including voice, music, or other audio data can be used. Audio payload is meant broadly to also include audio data included as part of video data. The process then proceeds to step 912. In step 912, the second audio stream's egress packets are then converted to cells. In one example, the cells are ATM cells. The process then proceeds to step 914. In step 914, cell switch 304 switches the cells to an SVC destined for the same destination NIC 306 on the same egress channel as the first audio stream. The process then proceeds to step 915. As shown in step 915 of FIG. 9B, SAR 634 now receives cells for the first and second audio streams. The cells are converted back to streams of internal egress packets and have control headers that include the respective priority information for the two audio streams. In step 916, NIC 306 compares the priorities of the two audio streams. If the second audio stream has a higher priority then the process proceeds to step 918. If not, then the process proceeds to step 930. In step 918, the transmission of the first audio stream is held. For example, NIC 306 buffers the first audio stream or even issues a control command to audio source 604a to hold the transmission of the first audio source. The process proceeds immediately to step 920. In step 920, the transmission of the second audio stream starts. NIC 306 instructs packet processor(s) 307 to create IP packets having the audio payload of the internal egress packets of the second audio stream. Packet processor(s) 307 add additional synchronized RTP header information (RTP packet information) and other header information (MAC, IP, UDP fields) to the audio payload of the internal egress packets of the second audio stream. NIC 306 then sends the IP packets with synchronized RTP header information on the same egress channel of the first audio stream. In this way, a destination device receives the second audio stream noise instead of the first audio stream. Moreover, from the perspective of the destination device this second audio stream is received in real-time noiselessly without delay or interruption. Steps 918 and 920 of course can be performed at the same time or in any order. The process proceeds immediately to step 922. As shown in FIG. 9C, NIC 306 monitors for the end of the second audio stream (step 922). The process proceeds immediately to step 924. In step 924, NIC 306 determines whether the second audio stream has ended. In one example, NIC 306 reads a last packet of the second audio stream which has a priority level lower than preceding packets. If so, then the process proceeds immediately to step 930. If not, then the process proceeds to step 922. In step 930, NIC 306 either continues to forward the first audio stream (after step 906) or returns to forwarding the first audio stream (after steps 916 or 924). The process proceeds to step 932. In one embodiment, NIC 306 maintains a priority level threshold value. NIC 306 then increments and sets the threshold based on priority information in the audio streams. When faced with multiple audio streams, NIC 306 forwards the audio stream having priority information equal to or greater than the priority level threshold value. For example, if the first audio stream had a priority value of 1 then the priority level threshold value is set to 1 and the first audio stream is transmitted (prior to step 904). When a second audio stream with a higher priority is received at NIC 306, then NIC 306 increments the priority threshold value to 2. The second audio stream is then transmitted as described above in step 920. When the last packet of the second audio stream having a priority field value set to 0 (or null or other special value) is read, then the priority level threshold value is decremented back to 1 as part of step 924. In this case, the first audio stream with priority information 1 is then be sent by NIC 306 as described above with respect to step 930. In step 932, egress audio controller 610 processes any remaining call events. The process then proceeds to step 934 where it terminates until re- instantiated. In one embodiment, the steps of the above-described process occur substantially at the same time, such that the process can be run in parallel or in an overlapping manner on one or more processors in the system 600. G. Audio Data Flow FIG. 6B is a diagram of audio data flow 615 in the noiseless switch over system of FIG. 6A in one embodiment. In particular, FIG. 6B shows the flow of internal packets from audio sources 604a-n to S ARs 630,632, the flow of cells through cell switch 304 to SAR 634, the flow of internal packets between SAR 634 and packet processors 307, and the flow of IP packets from NIC 306 over the network. H. Other Embodiments The present invention is not limited to internal audio sources or a cell layer. Noiseless switch over can also be carried out in different embodiments using internal audio sources only, internal and external audio sources, external audio sources only, a cell switch or a packet switch. For example, FIG. 6C is diagram of a noiseless switch over system 600C that carries out cell switching between independent egress audio streams generated by internal audio source 604a-n and/or external audio sources (not shown) according to an embodiment of the present invention. Noiseless switch over system 600C operates similar to system 600A described in detail above except that noiseless switch over is made to audio received from an external audio source. The audio is received in IP packets and buffered at NIC 306 as shown in FIG. 6C. NIC 306 strips IP information (stores it in forward table entry associated with external audio source and destination device) and generates internal packets assigned to a SVC. SAR 634 converts the internal packets to cells and routes cells on the SVC on link 662 through switch 304 back through link 664 to SAR 634 for conversion to internal packets. As described above, the internal packets are then processed by packet processor 307 to create IP packets with synchronized header information. NIC 306 then sends the IP packets to destination device. In this way, a user at the destination device is noiselessly switched over to receive audio from an external audio source. FIG. 6D is diagram of audio data flow 625 for an egress audio stream received from the external audio source in the noiseless switch over system of FIG. 6C. In particular, FIG. 6D shows the flow of IP packets from an external audio source (not shown) to NIC 306, the flow of internal packets from NIC 306 to SAR 634, the flow of cells through cell switch 304 back to SAR 634, the flow of internal packets between SAR 634 and packet processors 307, and the flow of IP packets from NIC 306 over the network to a destination device (not shown). FIG. 6E is diagram of audio data flows 635,645 in a noiseless switch over system 600E that carries out packet switching between independent egress audio streams generated by internal and/or externa] audio sources according to an embodiment of the present invention. Noiseless switch over system 600E operates similar to systems 600A and 600 C described in detail above except that a packet switch 694 is used instead of a cell switch 304. In this embodiment, a cell layer including SARs 630, 632, 634 is omitted. In audio data flow 635, internal packets flow through the packet switch 694 from internal audio sources 604a-n to packet processors 307. IP packets flow out to the network. In audio data flow 645, IP packets from an external audio source (not shown) are received at NIC 306. The audio is received in packets and buffered at NIC 306 as shown in FIG. 6E. NIC 306 strips IP information (stores it in forward table entry associated with external audio source and destination device) and generates internal packets assigned to a SVC (or other type of path) associated with the destination device. The internal packets are routed on the SVC through packet switch 694 to NIC 306. As described above, the internal packets are then processed by packet processor 307 to create IP packets with synchronized header information. NIC 306 then sends the IP packets to destination device. In this way, a user at the destination device is noiselessly switched over to receive audio from an external audio source. FIG. 6F is diagram of a noiseless switch over system 600F that carries out switching between independent egress audio streams generated by only external audio sources according to an embodiment of the present invention. No switch or internal audio sources are required. NIC 306 strips IP information (stores it in forward table entry associated with external audio source and destination device) and generates internal packets assigned to a SVC (or other type of path) associated with the destination device. The internal packets are routed on the SVC to NIC 306. (NIC 306 can be a common source and destination point). As described above, the internal packets are then processed by packet processor 307 to create IP packets with synchronized header information. NIC 306 then sends the IP packets to destination device. In this way, a user at the destination device is noiselessly switched over to receive audio from an external audio source. Functionality described above with respect to the operation of egress audio switching system 600 can be implemented in control logic. Such control logic can be implemented in software, firmware, hardware or any combination thereof. X. Conference Call Processing A. Distributed Conference Bridge FIG. 10 is a diagram of a distributed conference bridge 1000 according to one embodiment of the present invention. Distributed conference bridge 1000 is coupled to a network 1005. Network 1005 can be any type of network or combination of networks, such as, the Internet. For example, network 1005 can include a packet-switched network or a packeted-switched network in combination with a circuit-switched network. A number of conference call participants Cl-CN can connect through network 1005 to distributed conference bridge 1000. For example, conference call participants Cl-CN can place a VOIP call through network 1005 to contact distributed conference bridge 1000. Distributed conference bridge 1000 is scalable and can handle any number of conference call participants. For example, distributed conference bridge 1000 can handle conference calls between two conference call participants up to 1000 or more conference call participants. As shown in FIG. 10, distributed conference bridge 1000 includes a conference call agent 1010, network interface controller (NIC) 1020,switch 1030, and audio source 1040. Conference call agent 1010 is coupled to NIC 1020, switch 1030 and audio source 1040. NIC 1020 is coupled between network 1005 and switch 1030. Switch 1030 is coupled between NIC 1020 and audio source 1040. A look-up table 1025 is coupled to NIC 1020. Look-up table 1025 (or a separate look-up table not shown) can also be coupled to audio source 1040. Switch 1030 includes a muiticaster 1050. NIC 1020 includes a packet processor 1070. Conference call agent 1010 establishes a conference call for a number of participants. During a conference call, packets carrying audio, such as digitized voice, flow from the conference call participants Cl-CN to the conference bridge 1000. These packets can be IP packets including, but not limited to, RTP/RTCP packets. NIC 1020 receives the packets and forwards the packets along links 1028 to switch 1030. Links 1028 can be any type of logical and/or physical links such as PVCs or SVCs. In one embodiment, NIC 1020 converts IP packets (as described above with respect to FIG. 7A) to internal packets which only have a header and payload (as described with respect to FIG. 7B). The use of the internal packets further reduces processing work at audio source 1040. Incoming packets processed by NIC 1020 can also be combined by a SAR into cells, such as ATM cells, and sent over link(s) 1028 to switch 1030. Switch 1030 passes the incoming packets from NIC 1020 (or cells) to audio source J 040 on link(s) 1035. Link(s) 1035 can also be any type of logical and/or physical link including, but not limited to, a PVC or SVC. Audio provided over links 1035 is referred to in this conference bridge processing context as "external audio" since it originates from conference call participants over network 1005. Audio can also be provided internally through one or more links 1036 as shown in FIG. 10. Such "internal audio" can be speech, music, advertisements, news, or other audio content to be mixed in the conference call. The internal audio can be provided by any audio source or accessed from a storage device coupled to conference bridge 1000. Audio source 1040 mixes audio for the conference call. Audio source 1040 generates outbound packets containing the mixed audio and sends the packets over link(s) 1045 to switch 1030. In particular, audio source 1040 generates a fully mixed audio stream of packets and a set of partially mixed audio streams. In one embodiment, audio source 1040 (or "mixer" since it is mixing audio) dynamically generates the appropriate fully mixed and partially mixed audio streams of packets having conference identifier information (CID) and mixed audio during the conference call. The audio source retrieves the appropriate CED information of conference call participants from a relatively static look-up table (such as table 1025 or a separate table closer to audio source 1040) generated and stored at the initiation of the conference call. Multicaster 1050 multicasts the packets in the fully mixed audio stream and a set of partially mixed audio streams. In one embodiment, multicaster 1050 replicates the packets in each of the fully mixed audio stream and set of partially mixed audio streams N times which corresponds to the N number of conference call participants. The N replicated packets are then sent to endpoints in NIC 1020 over the N switched virtual circuits (SVC1-SVCN), respectively. One advantage of distributed conference bridge 1000 is that audio source 1040 (i.e., the mixing device) is relieved of the work of replication. This replication work is distributed to multicaster 1050 and switch 1030. NIC 1020 then processes outbound packets arriving on each SVC1-SVCN to determine whether to discard or forward the packets of the fully mixed and partially mixed audio streams to a conference call participant Cl-CN. This determination is made based on packet header information in real-time during a conference call. For each packet arriving on a SVC, NIC 1020 determines based on packet header information, such as TAS and IAS fields, whether the packet is appropriate for sending to a participant associated with the SVC. If yes, then the packet is forwarded for further packet processing. The packet is processed into a network packet and forwarded to the participant. Otherwise, the packet is discarded. In one embodiment, the network packet is an IP packet which includes the destination call participant's network address information (IP/UDP address) obtained from a look-up table 1025, RTP/RTCP packet header information (time stamp/sequence information), and audio data. The audio data is the mixed audio data appropriate for the particular conference call participant. The operation of distributed conference bridge 1000 is described further below with respect to an example look-up table 1025 shown in FIG. 11, flowchart diagrams shown in FIGs. 12 and 13A-13C, and example packet diagrams shown in FIGs. 14A, 14B and 15. B. Distributed Conference Bridge Operation FIG. 12 shows a routine 1200 for establishing conference bridge processing according to the present invention. (Steps 1200-1280). In step 1220, a conference call is initiated. A number of conference call participants Cl-CN dial distributed conference bridge 1000. Each participant can use any VOIP terminal including, but not limited to, a telephone, computer, PDA, set-top box, network appliance, etc. Conference call agent 1010 performs conventional IVR processing to acknowledge that a conference call participant wishes to participate in a conference call and obtains the network address of each conference call participant. For example, the network address information can include, but is not limited to, IP and/or UDP address information. In step 1240, look-up table 1025 is generated. Conference call agent 1010 can generate the look-up table or instruct NIC 1020 to generate the look-up table. As shown in the example on FIG. 11, look-up table 1025 includes N entries corresponding to the N conference call participants in the conference call initiated in step 1220. Each entry in look-up table 1025 includes an SVC identifier, conference ID (CID), and network address information. The SVC identifier is any number or tag that identifies a particular SVC. In one example, the SVC identifier is a Virtual Path Identifier and Virtual Channel Identifier (VPI/VCI). Alternatively, the SVC identifier or tag information can be omitted from look-up table 1025 and instead be inherently associated with the location of the entry in the table. For example, a first SVC can be associated with the first entry in the table, a second SVC can be associated with a second entry in the table, and so forth. The CID is any number or tag assigned by conference call agent 1010 to a conference call participant C1-CN. The network address information is the network address information collected by conference call agent 1010 for each of the N conference call participants. In step 1260, NIC 1020 assigns respective SVCs to each of the participants. For N conference call participants then N SVCs are assigned. Conference call agent 1010 instructs NIC 1020 to assign N SVCs. NIC 1020 then establishes N SVC connections between NIC 1020 and switch 1030. In step 1280, the conference call then begins. Conference call agent 1010 sends a signal to NIC 1020 and switch 1030 and audio source 1040 to begin conference call processing. Although FIG. 12 is described with respect to SVCs and SVC identifiers, the present invention is not so limited and any type of link (physical and/or logical) and link identifier can be used. Also, in embodiments where an internal audio source is included, conference call agent 1010 adds the internal audio source as one of the potential N audio participants whose input is to be mixed at audio source 1040. The operation of distributed conference bridge 1000 during conference call processing is shown in FIGs. I3A-13C (steps 1300-1398). Control begins at step 1300 and proceeds to step 1310. In step 1310, audio source 1040 monitors energy in the incoming audio streams of the conference call participant CI-CN. Audio source 1040 can be any type of audio source including, but not limited to, a digital signal processor (DSP). Any conventional technique for monitoring the energy of a digitized audio sample can be used. In step 1320, audio source 1040 determines a number of active speakers based on the energy monitored in step 1310. Any number of active speakers can be selected. In one embodiment, a conference call is limited to three active speakers at a given time. In this case, up to three active speakers are determined which correspond to the up to three audio streams having the most energy during the monitoring in step 1320. Next, audio source 1040 generates and sends fully mixed and partially mixed audio streams (steps 1330-1360). In step 1330, one fully mixed audio stream is generated. The fully mixed audio stream includes the audio content of the active speakers determined in step 1320. In one embodiment, the fully mixed audio stream is an audio stream of packets with packet headers and payloads. Packet header information identifies the active speakers whose audio content is included in the fully mixed audio stream. In one example, as shown in FIG. 14A audio source 1040 generates an outbound internal packet 1400 having a packet header 1401 with TAS. IAS, and Sequence fields and a payload 1403. The TAS field lists CIDs of all of the current active speaker calls in the conference call. The IAS field lists CIDs of the active speakers whose audio content is in the mixed stream. The sequence information can be a timestamp, numeric sequence value, or other type of sequence information. Other fields (not shown) can include checksum or other packet information depending upon a particular application. In the case of a fully mixed audio stream, the TAS and IAS fields are identical. Payload 1403 contains a portion of the digitized mixed audio in the fully mixed audio stream. In step 1340, audio source 1040 sends the fully mixed audio stream generated in step 1330 to switch 1030. Eventually, passive participants in the conference call (that is those determined not to be in the number of active speakers determined in step 1320), will hear mixed audio from the fully mixed audio stream. In step 1350, audio source 1040 generates a set of partially mixed audio streams. The set of partially mixed audio streams is then sent to switch 1030 (step 1360). Each of the partially mixed audio streams generated in step 1350 and sent in step 1360 includes the mixed audio content of the group of identified active speakers determined in step 1320 minus the audio content of a respective recipient active speaker. The recipient active speaker is the active speaker within the group of active speakers determined in step 1320 towards which a partially mixed audio stream is directed. In one embodiment, audio source 1040 inserts in packet payloads the digital audio from the group of identified active speakers minus the audio content of the recipient active speaker. In this way, the recipient active speaker will not receive audio corresponding to their own speech or audio input. However, the recipient active speaker will hear the speech or audio input of the other active speakers. In one embodiment, packet header information is included in each partially mixed audio stream to identify active speakers whose audio content is included in the respective partially mixed audio stream. In one example, audio source 1040 uses the packet format of FIG.14A and inserts one or more conference identification numbers (CIDs) into TAS and IAS header fields of packets. The TAS field lists CIDs of all of the current active speakers in the conference call. The IAS field lists CIDs of the active speakers whose audio content is in the respective partially mixed stream. In the case of a partially mixed audio stream, the TAS and IAS fields are not identical since the IAS field has one less CID. In one example, to build packets in steps 1330 and 1350, audio source 1040 retrieves the appropriate CID information of conference call participants from a relatively static look-up table (such as table 1025 or a separate table) generated and stored at the initiation of the conference call. For example, in a conference call where there are 64 participants (N = 64) of which three are identified as active speakers (1-3), then one fully mixed audio stream will contain audio from all three active speakers. This fully mixed stream is eventually sent to each of the 61 passive participants. Three partially mixed audio streams are then generated in step 1350. A first partially mixed stream 1 contains audio from speakers 2-3 but not speaker 1. A second partially mixed stream 2 contains audio from speakers 1-3 but not speaker 2. A third partially mixed stream 3 contains audio from speakers 1 and 2 but not speaker 3. The first through third partially mixed audio streams are eventually sent to speakers 1-3 respectively. In this way only four mixed audio streams (one fully mixed and three partially mixed) need be generated by audio source 1040. This reduces the work on audio source 1040. As shown in FIG. 13B, in step 1370, multicaster 1050 replicates packets in the fully mixed audio stream and set of partially mixed audio streams and multicasts the replicated packet copies on all of the SVCs (SVC1-SVCN) assigned to the conference call. NIC 1020 then processes each packet received on the SVC (step 1380). For clarity, each packet processed internally in distributed conference bridge 1000 (including packets received at SVCs by NIC 1020) are referred to as internal packets. Internal packets can be any type of packet format including, but not limited to, IP packets and/or internal egress packets described above in Figs. 7A and 7B, and the example internal egress or outbound packet described with respect to FIG. 14A. For each SVC, NIC 1020 determines whether to discard or forward a received internal packet for further packet processing and eventual transmission to a corresponding conference call participant (step 1381). The received internal packet can be from a fully mixed or partially mixed audio stream. If yes, the packet is to be forwarded, then control proceeds to step 1390. If no, the packet is not to be forwarded, then control proceeds to step 1380 to process the next packet. In step 1390, the packet is processed into a network IP packet. In one embodiment, packet processor 1070 generates a packet header with at least the participant's network address information (IP and/or UDP address) obtained from the look-up table 1025. Packet processor 1070 further adds sequence information such as RTP/RTCP packet header information (e.g., a timestamp and/or other type of sequence information). Packet processor 1070 can generate such sequence information based on the order of received packets and/or based on sequence information (e.g. the Sequence field) provided in packets generated by the audio source 1040 (or by multicaster 1050). Packet processor 1070 further adds a payload in each network packet that includes audio from the received internal packet being forwarded to a participant. NIC 1020 (or packet processor 1070) then sends the generated IP packet to the participant (step 1395). One feature of the present invention is that the packet processing determination in step 1381 can be performed quickly and in real-time during a conference call. FIG. 13C shows one example routine for carrying out the packet processing determination step 1381 according to the present invention (steps 1382-1389). This routine is carried out for each outbound packet that arrives on each SVC. NIC 1020 acts as a filter or selector in determining which packets are discarded and which are converted to IP packets and sent to a call participant. When an internal packet arrives on a SVC, NIC 1020 looks up an entry in look up table 1025 that corresponds to the particular SVC and obtains a CID value (step 1382). NIC 1020 then determines whether the obtained CID value matches any CID value in the Total Active Speakers (TAS) field of the internal packet (step 1383). If yes, control proceeds to step 1384. If no, control proceeds to step 1386. In step 1384, NIC 1020 determines whether the obtained CID value matches any CID value in the Included Active Speakers (IAS) field of the internal packet. If yes, control proceeds to step 1385. If no, control proceeds to step 1387. In step 1385, the packet is discarded. Control then proceeds to step 1389 which returns control to step 1380 to process a next packet. In step 1387, control jumps to step 1390 for generating an IP packet from the internal packet. In step 1386, a comparison of the TAS and IAS fields is made. If the fields are identical (as in the case of a fully mixed audio stream packet), then control proceeds to step 1387. In step 1387, control jumps to step 1390. If the TAS and IAS fields are not identical, then control proceeds to step 1385 and the packet is discarded. C. Outbound Packet Flow through Distributed Conference Bridge Outbound packet flow in distributed conference bridge 1000 is described further with respect to example packets in a 64-person conference call shown in FIGs. 14 and 15. In FIGs. 14 and 15, mixed audio content in a packet payload is denoted by a bracket surrounding the respective participants whose audio is mixed (e.g., {C1,C2,C3}). CID information in packet headers is denoted by underlining the respective active speaker participants (e.g., C1, C2, C3, etc.). Sequence information is simply shown by a sequence number 0,1 etc. In this example, there are 64 participants C1-C64 in a conference call of which three are identified as active speakers at a given time (C1-C3). Audio participants C4-C64 are considered passive and their audio is not mixed. Audio source 1040 generates one fully mixed audio stream FM having audio from all 3 active speakers (C1-C3). FIG. 14B shows two example internal packets 1402, 1404 generated by audio source 1040 during this conference call. Packets 1402, 1404 in stream FM have a packet header and payload. The payloads in packets 1402,1404 each include mixed audio from each of the three active speakers C1- C3. Packets 1402,1404 each include packet headers having TAS and IAS fields. The TAS field contains CIDs for the total three active speakers C1-C3. The IAS field contains CIDs for the active speakers C1-C3 whose content is actually mixed in the payload of the packet. Packet 1402,1404 further include sequence information 0 and 1 respectively to indicate packet 1402 precedes packet 1404. Mixed audio from fully mixed stream FM is eventually sent to each of the 61 currently passive participants (C4-C64). Three partially mixed audio streams PM1-PM3 are generated by audio source 1040. FIG. 14B shows two packets 1412, 1414 of first partially mixed stream PM1. Payloads in packets 1412 andl414 contain mixed audio from speakers C2 and C3 but not speaker CI. Packets 1412,1414 each include packet headers. The TAS field contains CIDs for the total three active speakers C1-C3. The IAS field contains CIDs for the two active speakers C2 and C3 whose content is actually mixed in the payload of the packet. Packet 1412,1414 have sequence information 0 and 1 respectively to indicate packet 1412 precedes packet 1414. FIG. 14B shows two packets 1422,1424 of second partially mixed stream PM2. Payloads in packets 1422 and 1424 contain mixed audio from speakers C1 and C3 but not speaker C2. Packets 1422,1424 each include packet headers. The TAS field contains CIDs for the total three active speakers C1-C3. The IAS field contains CIDs for the two active speakers C1 and C3 whose content is actually mixed in the payload of the packet. Packets 1422,1424 have sequence information 0 and 1 respectively to indicate packet 1422 precedes packet 1424. FIG. 14B further shows two packets 1432,1434 of third partially mixed stream PM3. Payloads in packets 1432 and 1434 contain mixed audio from speakers C1 and C2 but not speaker C3. Packets 1432,1434 each include packet headers. The TAS field contains CIDs for the total three active speakers C1-C3. The IAS field contains CIDs for the two active speakers C1 and C2 whose content is actually mixed in the payload of the packet. Packets 1432,1434 have sequence information 0 and 1 respectively to indicate packet 1432 precedes packet 1434. FIG. 15 is a diagram that illustrates example packet content after the packets of FIG. 14 have been multicasted and after they have been processed into IP packets to be sent to appropriate conference call participants according to the present invention. In particular, packets 1412,1422,1432,1402,1414 are shown as they are multicast across each of SVC1-SVC64 and arrive at NIC 1020. As described above with respect to step 1381, NIC 1020 determines for each SVC1- SVC64 which packets 1412,1422,1432,1402,1414 are appropriate to forward to a respective conference call participant C1-C64. Network packets (e.g. IP packets) are then generated by packet processor 1070 and sent to the respective conference call participant C1-C64. As shown in FIG. 15, for SVC1, packets 1412 and 1414 are determined to be forwarded to C1 based on their packet headers. Packets 1412,1414 have the CID of C1 in the TAS field but not the IAS field. Packets 1412 and 1414 are converted to network packets 1512 and 1514. Network packets 1512, 1514 include the IP address of C1 (C1ADDR) and die mixed audio from speakers C2 and C3 but not speaker CI. Packets 1512,1514 have sequence information 0 and 1 respectively to indicate packet 1512 precedes packet 1514. For SVC2 (corresponding to conference call participant C2), packet 1422 is determined to be forwarded to C2. Packet 1422 has the CID of C2 in the TAS field but not the IAS field. Packet 1422 is converted to network packet 1522. Network packet 1522 includes the IP address of C2 (C2ADDR), sequence information 0, and the mixed audio from speakers C1 and C3 but not speaker C2. For SVC3 (corresponding to conference call participant C3), packet 1432 is determined to be forwarded to C3. Packet 1432 has the CID of C3 in the TAS field but not the IAS field. Packet 1432 is converted to network packet 1532. Network packet 1532 includes the IP address of C3 (C3ADDR), sequence information 0, and the mixed audio from speakers C1 and C2 but not speaker C3. For SVC4 (corresponding to conference call participant C4), packet 1402 is determined to be forwarded to C4. Packet 1402 does not have the CID of C4 in the TAS field and the TAS and IAS fields are identical indicating a fully-mixed stream. Packet 1402 is converted to network packet 1502. Network packet 1502 includes the IP address of C4 (C4ADDR), sequence information 0, and the mixed audio from all of the active speakers C1, C2, and C3. Each of the other passive participants C5- C64 receive similar packets. For example, for SVC64 (corresponding to conference call participant C64), packet 1402 is determined to be forwarded to C64. Packet 1402 is converted to network packet 1503. Network packet 1503 includes the IP address of C64 (C64ADDR), sequence infonnation 0, and the mixed audio from all of the active speakers C1, C2, and C3. D. Control Logic and Additional Embodiments Functionality described above with respect to the operation of conference bridge 1000 (including conference call agent 1010, NIC 1020, switch 1030, audio source 1040, and multi-caster 1050) can be implemented in control logic. Such control logic can be implemented in software, firmware, hardware or any combination thereof. In one embodiment, distributed conference bridge 1000 is implemented in a media server such as media server 202. In one embodiment, distributed conference bridge 1000 is implemented in audio processing platform 230. Conference call agent 1010 is part of call control and audio feature manager 302. NIC 306 carries out the network interface functions of NIC 1020 and packet processors 307 carry out the function of packet processor 1070. Switch 304 is replaced with switch 1030 and multicaster 1050. Any of audio sources 308 can carry out the function of audio source 1040. XI. Conclusion While specific embodiments of the.present invention have been described above, it should be understood that they have been presented by way of example only, and not limitation. It will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention as defined in the appended claims. Thus, the breadth and scope of the present invention should not be limited by any of the above-described exemplary embodiments, but should be defined only in accordance with the following claims and their equivalents. WE CLAIM : 1. 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 Tn the respective packets. 2. The method as claimed in claim 1, 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 theJnitiated conference call. 3. The method as claimed in claim 2, comprising: assigning switched virtual circuits (SVCs) to respective participants in the conference call; wherein said storing step also comprises 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. 4. The method as claimed in claim 1, comprising: monitoring energy in inbound audio streams of the participants; and determining a number of active speakers based on the monitored energy. 5. The method as claimed in claim 4, wherein said generating steps (a) and (b) generate packet headers having active speaker information based on the determined member of active speakers, and said determining step (d) determines which malticasted packets to forward to participants based on the active speaker information in the respective packet headers. 6. The method as claimed in claim 5, 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. 7. The method as claimed in claim 6, 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. 8. The method as claimed in claim 6, wherein said determining step (d) comprises for each packet being processed at a SVC the steps of: obtaining a CID value for the SVC; and 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. 9. The method as claimed in claim 8, 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. 10. The method as claimed in claim 6, wherein the packet header comprises 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. 11. The method as claimed in claim 1, 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. 12. The method as claimed in claim 1, 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. 13. The method as claimed in claim 1, comprising mixing audio received over a network from participants in the conference call who are active speakers. 14. The method as claimed in claim 1, 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. 15. 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 pay load; a switch; and a network interface controller, wherein said switch is coupled between said network interface controller and said audio source and said switch also 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. 16. The conference bridge as claimed in claim 15, comprising a conference call agent that initiates a conference call among the participants. 17. The conference bridge as claimed in claim 16, comprising a storage device that stores conference identifier.information. (CID) and network address information, associated with each participant in the established conference call. 18. The conference bridge as claimed in claim 17, wherein said storage device comprises a look-up table. 19. The conference bridge as claimed in claim 17, 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. 20. The conference bridge as claimed in claim 15, 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. 21. The conference bridge as claimed in claim 20, 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. 22. The conference bridge as claimed in claim 21, 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. 23. The conference bridge as claimed in claim 22, wherein said packet headers generated by said audio source also comprise sequence information. 24. The conference bridge as claimed in claim 15, 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.jn. said set have payloads which include mixed audio from the at least three active speakers minus the audio of a respective recipient active speaker. 25. The conference bridge as claimed in claim 15, comprising: a packet processor that processes packets determined to be forwarded into network packets having network addresses of participants in the conference call. 26. The conference bridge as claimed in claim 15, wherein said audio source mixes audio received over a network from participants in the conference call who are active speakers. 27. The conference bridge as claimed in claim 15, 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. 28. 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. 29. A media server for use in a VOIP 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 also 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. 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.

Full Text

METHOD AND SYSTEM FOR PROCESSING AUDIO IN A
CONFERENCE CALL AMONG PARTICIPANTS
BACKGROUND OF THE INVENTION
Field of the Invention
The invention relates generally to audio communication over a network.
Background Art
Audio has long been carried in telephone calls over networks. Traditional
circuit-switched time division multiplexing (TDM) networks including public-
switched telephone networks (PSTN) and plain old telephone networks (POTS)
were used. These circuit-switched networks establish a circuit across the network
for each call. Audio is carried in analog and/or digital form across the circuit in
real-time.
The emergence of packet-switched networks, such as the local area
networks (LANs), and the Internet, now requires that audio be carried digitally
in packets. Audio can include but is not limited to voice, music, or other type of
audio data. Voice over Internet Protocol systems (also called Voice over IP or
VOIP systems) transport the digital audip data belonging to a telephone call in
packets over packet-switched networks instead of traditional circuit-switched
networks. In one example, a VOIP system forms two or more connections using
Transmission Control Protocol/Internet Protocol (TCP/IP) addresses to
accomplish a connected telephone call. Devices that connect to a VOIP network
must follow standard TCP/IP packet protocols in order to intemperate with other
devices within the VOIP network. Examples of such devices are IP phones,
integrated access devices, media gateways, and media servers.
A media server is often an endpoint in a VOIP telephone call. The media
server is responsible for ingress and egress audio streams, that is, audio streams
which enter and leave a media server respectively. The type of audio produced
by a media server is controlled by the application that corresponds to the
telephone call such as voice mail, conference bridge, interactive voice response
(IVR), speech recognition, etc. In many applications, the produced audio is not
predictable and must vary based on end user responses. Words, sentences, and
whole audio segments such as music must be assembled dynamically in real time
as they are played out in audio streams.
Packet-switched networks, however, can impart delay and jitter in a
stream of audio carried in a telephone call. A real-time transport protocol (RTP)
is often used to control delays, packet loss and latency in an audio stream played
out of a media server. The audio stream can be played out using RTP over a
network link to a real-time device (such as a telephone) or a non-real-time device
(such as an email client in unified messaging). RTP operates on top of a protocol
such as the User Datagram Protocol (UDP) which is part of the IP family. RTP
packets include among other things a sequence number and a timestamp. The
sequence number allows a destination application using RTP to detect the
occurrence of lost packets and to ensure a correct order of packets are presented
to a user. The timestamp corresponds to the time at which the packet was
assembled. The timestamp allows a destination application to ensure
synchronized play-out to a destination user and to calculate delay and jitter. See,
D. Collins, Carrier Grade Voice over IP,Mc-Grsw Hill: United States, Copyright
2001, pp. 52-72, the entire book of which is incorporated in its entirety herein by reference.
A media server at an endpoint in a VOIP telephone call uses protocols
such as RTP to improve communication quality for a single audio stream. Such
media servers, however, have been limited to outputting a single audio stream of
RTP packets for a given telephone call.
A conference call links multiple parties over a network in a common call.
Conference calls were originally carried out over a circuit-switched network such
as a plain old telephone system (POTS) or public switched telephone network
(PSTN). Conference calls are now also carried out over packet-switched
networks, such as local area networks (LANs) and the Internet. Indeed, the
emergence of voice over the Internet systems (also called Voice over IP or VOIP
systems) has increased the demand for conference calls over networks.
Conference bridges connect participants in conference calls. Different
types of conference bridges have been used depending in part upon the type of
network and how voice is carried over the network to the conference bridge. One
type of conference bridge is described in U.S. Pat. No. 5,436,896 (see the entire
patent). This conference bridge 10 operates in an environment where voice
signals are digitally encoded in a 64Kbps data stream (FIG. 1, col. 1, Ins. 21-26).
Conference bridge 10 has a plurality of inputs 12 and outputs 14. Inputs 12 are
connected through respective speech detectors 16 and switches 18 to a common
summing amplifier 20. Speech detector 16 detects speech by sampling an input
data stream and determining the amount of energy present over time. (col. 1, Ins.
36-39). Each speech detector 16 controls a switch 18. When no speech is present
switch 18 is held open to reduce noise. During a conference call, inputs 12 of all
participants who are speaking are coupled through summing amplifier 20 to each
of the outputs 14. Subtracters 24 subtract each participant's own voice data
stream. A number of participants 1-n then can speak and hear each other in the
connections made through conference bridge 10. See, '896 patent, col. 1, In, 12-
col. 2, In. 16.
Digitized voice is now also being carried in packets over packet-switched
networks. The '896 patent describes one example of asynchronous mode transfer
(ATM) packets (also called cells). To support a conference call in this
networking environment, conference bridge 10 converts input ATM cells to
network packets. Digitized voice is extracted from the packets and processed in
conference bridge 12 as described above. The summed output digitized voices
is re-converted from network packets back to ATM cells prior to being sent to
participants 1-n. See, '896 patent, col. 2, In. 17-col. 2, In. 36.
The '896 patent also describes a conference bridge 238 shown in FIGs. 2
and 3 which processes ATM cells without converting and re-converting the ATM
cells to network packets as in conference 10. Conference bridge 238 has inputs
302-306, one from each of the participants, and outputs 308-312, one to each of
the participants. Speech detectors 314-318 analyze input data aggregated in
sample and hold buffers 322-326. Speech detectors 314-318 report the detected
speech an/or volume of detected speech to controller 320. See, 4896 patent, col.
4, Ins. 26-39.
Controller 320 is coupled to a selector 328, gain control 329 and replicator
330. Controller 320 determines which of the participants is speaking based on
the outputs of speech detectors 314-318. When one speaker (such as participant
1) is talking, controller 320 sets selector 328 to read data from buffer 322. The
data moves through automatic gain control 329 to replicator 330. Replicator
replicates the data in the ATM cell selected by selector 328 for all participants
except the speaker. See, '896 patent, col. 4, In. 40-col. 5, In. 5. When two or
more speakers are speaking, the loudest speaker is selected in a given selection
period. The next loudest speaker is then selected in a subsequent selection
period. The appearance of simultaneous speech is kept up by scanning speech
detectors 314-318 and reconfiguring selector 328 at appropriate interval such as
six milliseconds. See, '896 patent, col. 5, Ins. 6-65.
Another type of conference bridge is described in U.S. Pat. No. 5,983,192
(see the entire patent). In one embodiment, a conference bridge 12 receives
compressed audio packets through a real-time transport protocol (RTP/RTCP).
See, '192 patent, col. 3, In. 66-col. 4, In. 40. Conference bridge 12 includes
audio processors 14a-14d. Exemplary audio processor 14c associated with a site
C (i.e., a participant C) includes a switch 22 and selector 26. Selector 26 includes
a speech detector which determines which of other sites A, B, or D has the
highest likelihood of speech. See, '192 patent, col. 4, Ins. 40-67. Alternatives
include selecting more than one site and using an acoustic energy detector. See,
' 192 patent, col. 5, Ins. 1-7. In another embodiment described in the' 192 patent,
the selector 26/switches 22 output a plurality of loudest speakers in separate
streams to local mixing end-point sites. The loudest streams are sent to multiple
sites. See, '192 patent, col. 5, Ins. 8-67. Configurations of mixer/encoders are
also described to handle multiple speakers at the same time, referred to as
"double-talk" and "triple-talk." See, '192 patent, col. 7, In. 20-col. 9, In. 29.
Voice-over-the-Intemet (VOIP) systems continue to require an improved
conference bridge. For example, a Softswitch VOIP architecture may use one or
more media servers having a media gateway control protocol such as MGCP
(RFC 2705). See, D. Collins, Carrier Grade Voice over IP, Mc-Graw Hill:
United States, Copyright 2001, pp. 234-244, the entire book of which is
incorporated in its entirety herein by reference. Such media servers are often used
to process audio streams in VOIP calls. These media servers are often endpoints
where audio streams are mixed in a conference call. These endpoints are also
referred to as "conference bridge access points" since the media server is an
endpoint where media streams from multiple callers are mixed and provided
again to some or all of the callers. See, D. Collins, p. 242.
As the popularity and demand for IP telephony and VOIP calls increases,
media servers are expected to handle conference call processing with carrier
grade quality. Conference bridges in a media server need to be able to scale to
handle different numbers of participants. Audio in packet streams, such as
RTP/RTCP packets, needs to be processed in real-time efficiently.
BRIEF SUMMARY OF THE INVENTION
The present invention provides a method and system for providing media
services in Voice over IP telephony. In one embodiment, a switch is coupled
between multiple audio sources and a network interface controller. The switch
can be a packet switch or a cell switch. Internal and/or external audio sources
generate audio streams of packets. Any type of packet can be used. In one
embodiment, an internal packet includes a packet header and a payload.
In one embodiment, the packet header has information that identifies
active speakers whose audio is mixed. The payload carries the digitized mixed
audio. According to a feature of the present invention, one fully mixed audio
stream of packets and a number of partially mixed audio streams of packets are
generated by an audio source (e.g. a DSP). The fully mixed audio stream
includes the audio content of a group of identified active speakers. Packet header
information identifies each of the active speakers in the fully mixed stream. In
one example, the audio source inserts conference identification numbers (CIDs)
associated with the respective active speakers into header fields in the packets.
The audio source inserts mixed digital audio from the active speakers into the
payload of the packets. The mixed digital audio corresponds to speech or other
type of audio that is input by the active speakers in the conference call.
Each of the partially mixed audio streams includes the audio content of
the group of identified active speakers minus the audio content of a respective
recipient active speaker. The recipient active speaker is the active speaker within
the group of active speakers towards which a partially mixed audio stream is
directed. The audio source inserts into the packet payloads the digital audio from
the group of identified active speakers minus the audio content of the recipient
active speaker. In this way, the recipient active speaker will not receive audio
corresponding to their own speech or audio input. Packet header information
identifies the active speakers whose audio content is included in the respective
partially mixed audio stream. In one example, the audio source inserts one or
more conference identification numbers (CIDs) into TAS and IAS header fields
of packets. The TAS (Total Active Speakers) field lists CIDs of all of the current
active speaker calls in the conference call. The IAS field (Included Active
Speakers) lists CIDs of the active speakers whose audio content is in the
respective partially mixed stream. In one embodiment, the audio source (or
"mixer" since it is mixing audio) dynamically generates the appropriate fully
mixed and partially mixed audio streams of packets having CDD information and
mixed audio during the conference call. The audio source retrieves the
appropriate CID information of conference call participants from a relatively
static look-up table generated and stored at the initiation of the conference call.
For example, in a conference call where there are 64 participants of which
3 are identified as active speakers (1-3), then one fully mixed audio stream will
contain audio from all 3 active speakers. This fully mixed stream is eventually
sent to each of the 61 passive participants. Three partially mixed audio streams
are also generated. A first partially mixed stream 1 contains audio from speakers
2-3 but not speaker 1. A second partially mixed stream 2 contains audio from
speakers 1-3 but not speaker 2. A third partially mixed stream 3 contains audio
from speakers 1 -2 but not speaker 3. The first through third partially mixed audio
streams are eventually sent to speakers 1-3 respectively. In this way only four
mixed audio streams need be generated by the audio source.
The fully mixed audio stream and a number of partially mixed audio
streams are sent from the audio sources (e.g. DSPs) to a packet switch. A cell
layer can also be used. The packet switch multicasts each fully mixed and
partially mixed audio stream to a network interface controller (NIC). The NIC
then processes each packet to determine whether to forward the packet for the
fully mixed or partially mixed audio stream to a participant. This determination
can be made in real-time based on a look-up table at the NIC and the packet
header information in the multicasted audio streams.
In one embodiment, during initialization of the conference call, each
participant in the call is assigned a CID. A switched virtual circuit (SVC) is also
associated with the conference call participant. A look-up table is generated and
stored which includes entries for the conference call participants. Each entry
includes the network address information (e.g, IP, UDP address information) and
CID of a respective conference call participant. Look-up tables can be stored for
access by both the NIC processing packets and the audio source(s) mixing audio
during a conference call.
The packet switch multicasts each fully mixed and partially mixed audio
stream on all of the SVCs assigned to the conference call to the NIC. The NIC
processes each packet arriving on the SVCs and in particular examines packet
headers to determine whether to discard or forward the packet for the fully mixed
or partially mixed audio stream to a participant. One advantage of the present
invention is that this packet processing determination can be performed quickly
and in real-time during a conference call based on packet header information and
CID information obtained from the lookup table. In one embodiment, the
network packet which is sent includes the participant's network address
information (IP/UDP) obtained from the look-up table, RTP packet header
information (timestamp/sequence information), and audio data.
In sum, advantages of the present invention include providing conference
bridge processing using fewer resources with less bandwidth and processing than
is typically required in mixing devices in other conference bridges. A conference
bridge system and method of the present invention multicasts in a way that
relieves the mixing device of the work of replication. For a conference call with
N participants and c active speakers, an audio source only needs to generate c +
1 mixed audio streams (one fully mixed audio stream and c partially mixed audio
streams). Work is distributed to the a multicaster in a switch which performs
replication and multicasts the mixed audio streams. A further advantage is that
a conference bridge according to the present invention can scale to accommodate
large numbers of participants. For example, if N = 1000 participants, and c = 3
active speakers then an audio source only need generate c + 1 = 4 mixed audio
streams. Packets in multicasted audio streams are processed at a NIC in real-time
to determine the appropriate packets for output to a participant in the conference
call. In one example, internal egress packets having a header and payload are
used in the conference bridge to further reduce processing work at an audio
source mixing the audio for the conference call.
In addition, as the use of audio networking increases and the number of
users and applications rise, there is an increasing need for multiple audio streams
even in a given telephone call. The inventors recognized that multiple audio
streams need to be switched dynamically without introducing RTP errors in calls
placed in an audio networking environment such as a voice over IP network.
Such RTP errors can cause unwanted noises such as clicks, pops, etc.
The present invention 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. For established
VOIP calls, the present invention can noiselessly switch audio from one audio
source to another. This switching system is dynamic and can scale to handle
many calls.
In embodiments of the present invention, a switch is used to direct audio
data from multiple audio sources to a network interface controller. The switch
can be a cell switch or a packet switch. The audio sources can be internal audio
sources and/or external audio sources. The network interface controller (NIC)
can be any interface with an IP network and includes one or more packet
processors. An egress audio controller controls the operation of internal audio
sources, the switch and the network interface controller to carry out noiseless
switching according to the present invention.
In one feature of the invention, priority information is used by a network
interface controller to determine which audio stream from an internal or external
audio source is transmitted in an established VOIP telephone call. Consider the
case of two internal audio sources. The audio sources generate respective audio
streams of internal egress packets for one destination egress audio channel. In
one embodiment, each internal egress packet includes a payload carrying audio
and control header information. The control header information has priority
information. This priority information is then used by a network interface
controller to determine which audio stream is transmitted because only one RTP
stream can be output at a given time for each VOIP call.
In one feature of the invention, the internal egress packets are smaller than
IP packets and consist of payload and control header information only. In this
way, processing work required to create complete IP packets need not be carried
out by internal audio sources such as DSPs but is distributed to the packet
processors in the network interface controller.
According to further feature, a cell switch is used which is a fully meshed
cell switch such as an ATM cell switch that has plenty of available bandwidth.
The internal egress packets for the different audio streams are converted to cells.
The cell switch combines merged cells from different sources and delivers them
across a switched virtual circuit (SVC) to a NIC. The SVC is associated with one
egress output audio channel serving an established telephone call.
In one embodiment, an egress audio controller is used to control noiseless
switching of audio in VOIP telephone calls. This noiseless switching according
to the present invention is also referred to herein as a "noiseless switch over." In
one embodiment, noiseless switch over of additional audio is carried out for calls
in which this service is available. In this way, an extra charge may be made for
providing a noiseless switch over service. In other embodiments, noiseless
switch over is performed for any call.
Certain call events which involve additional audio trigger the noiseless
switch over. This noiseless switch over is carried out using the noiseless
switching system and method of the present invention. Examples of call events
include but are not limited to the following conditions: an emergency condition,
a call signaling condition, a call event based on callee or caller information, or a
request for different audio information. The request for audio information can be
any audio request such as a request for advertisements, news sports, financial,
music or other audio content.
Audio sources can generate any type of audio. For example, an audio
stream of egress packets can include audio payloads representing voice, music,
tones, and/or any other sound.
The egress audio controller can be a stand-alone unit or a part of a call
control and audio feature manager in an audio processing platform. The present
invention can be implemented in a media server, audio processor, router, packet
switch, or audio processing platform.
Another embodiment involves the switching of audio streams including
an audio stream from an external audio source. In this case, a NIC receives IP
packets containing the audio stream and converts the IP packets to internal egress
packets. At this point, the internal egress packets are processed as if they were
generated by an internal audio source. The internal egress packets may include
priority information. The internal egress packets may be sent as packets or cells
across a SVC through a switch to the NIC. When the external audio stream has
a relatively high priority and switch over is to proceed, a packet processor at the
NIC generates IP packets with synchronized header information (such as RTP
information) and sends the IP packets to a destination device.
In one embodiment, a noiseless switch over system according to the
invention involves the switching of audio streams only from internal audio
sources such as DSPs. In another embodiment, a noiseless switch over system
according to the invention involves the switching of audio streams from internal
audio sources and external audio sources. In another embodiment, a noiseless
switch over system according to the invention involves the switching of audio
streams only from external audio sources in which case the switch over system
acts a general switch for audio streams and no internal DSPs are required.
Further embodiments, features, and advantages of the present inventions,
as well as the structure and operation of the various embodiments of the present
invention, are described in detail below with reference to the accompanying
drawings.
BRIEF DESCRIPTION OF THE ACCOMPANYING DRAWINGS
The accompanying drawings, which are incorporated herein and form a
part of the specification, illustrate the present invention and, together with the
description, further serve to explain the principles of the invention and to enable
a person skilled in the pertinent art to make and use the invention.
In the drawings:
FIG. 1 is a diagram of a media server in a voice over the Internet example
environment according to the present invention.
FIG. 2 is a diagram of an example media server including media services
and resources according to the present invention.
FIGs. 3 A and 3B are diagrams of an audio processing platform according
to an embodiment of the present invention.
FIG. 4 is a diagram of a audio processing platform as shown in FIG. 3
according to an example implementation of the present invention.
FIG. 5A is a flow diagram showing the establishment of a call and ingress
packet processing according to an embodiment of the present invention.
FIG. 5B is a flow diagram showing egress packet processing and call
completion according to an embodiment of the present invention.
FIGs. 6A-6F are diagrams of noiseless switch over systems according to
embodiments of the present invention.
FIG. 6A is diagram of a noiseless switch over system that carries out cell
switching of independent egress audio streams generated by internal audio
sources according to an embodiment of the present invention.
FIG. 6B is diagram of audio data flow in a noiseless switch over system
that carries out cell switching of independent egress audio streams generated by
internal audio sources according to an embodiment of the present invention.
FIG. 6C is diagram of a noiseless switch over system that carries out cell
switching between independent egress audio streams generated by internal and/or
external audio sources according to an embodiment of the present invention.
FIG. 6D is diagram of audio data flow in a noiseless switch over system
that carries out cell switching between independent egress audio streams
generated by internal and/or external audio sources according to an embodiment
of the present invention.
FIG. 6E is diagram of audio data flow in a noiseless switch over system
that carries out packet switching between independent egress audio streams
generated by internal and/or external audio sources according to an embodiment
of the present invention.
FIG. 6F is diagram of a noiseless switch over system that carries out
switching between independent egress audio streams generated by external audio
sources according to an embodiment of the present invention.
FIG. 7A is a schematic illustration of an IP packet with RTP information.
FIG. 7B is a schematic illustration of an internal packet according to one
embodiment of the present invention.
FIG. 8 is a flow diagram showing the switching functionality according
to one embodiment of the present invention.
FIG. 9A, 9B, and 9C are flow diagrams showing the call event processing
for audio stream switching according to one embodiment of the present invention.
FIG. 10 is a block diagram of a distributed conference bridge according
to one embodiment of the present invention.
FIG. 11 is an example look-up table used in the distributed conference
bridge of FIG. 10.
FIG. 12 is a flowchart diagram of the operation of the distributed
conference bridge of FIG. 10 in establishing a conference call.
FIGs. 13A, 13B, and 13C are flowchart diagrams of the operation of the
distributed conference bridge of FIG. 10 in processing a conference call.
FIG. 14A is a diagram of an example internal packet generated by an
audio source during a conference call according to one embodiment of the present
invention.
FIG. 14B is a diagram that illustrates example packet content in a fully
mixed audio stream and set of partially mixed audio streams according to the
present invention.
FIG. 15 is a diagram that illustrates example packet content after the
packets of FIG. 14 have been multicasted and after they have been processed into
IP packets to be sent to appropriate participants in a 64 participant conference call
according to the present invention.
The present invention will now be described with reference to the
accompanying drawings. In the drawings, like reference numbers indicate
identical or functionally similar elements. Additionally, the left-most digit(s) of
a reference number identifies the drawing in which the reference number first
appears.
DETAILED DESCRIPTION OF THE INVENTION
I. Overview and Discussion
The present invention provides a method and system for distributed
conference bridge processing in Voice over IP telephony. Work is distributed
away from a mixing device such as a DSP. In particular, a distributed conference
bridge according to the present invention uses internal multicasting and packet
processing at a network interface to reduce work at an audio mixing device. A
conference call agent is used to establish and end a conference call. An audio
source such as a DSP mixes audio of active conference call participants. Only
one fully mixed audio stream and a set of partially mixed audio streams need to
be generated. A switch is coupled between the audio source mixing audio content
and a network interface controller. The switch includes a multi-caster. The
multi-caster replicates packets in the one fully mixed audio stream and a set of
partially mixed audio streams and multi-casts the replicated packets to links (such
as SVCs) associated with each call participant. A network interface controller
processes each packet to determine whether to discard or forward the packet for
the fully mixed or partially mixed audio stream to a participant. This
determination can be made in real-time based on a look-up table at the NIC and
the packet header information in the multicasted audio streams.
In one embodiment, a conference bridge according to the present
invention is implemented in a media server. According to embodiments of the
present invention, the media server can include a call control and audio feature
manager for managing the operations of the conference bridge.
The present invention is described in terms of an example voice over the
Internet environment. Description in these terms is provided for convenience
only. It is not intended that the invention be limited to application in these
example environments. In fact, after reading the following description, it will
become apparent to a person skilled in the relevant art how to implement the
invention in alternative environments known now or developed in the future.
II. Terminology
To more clearly delineate the present invention, an effort is made
throughout the specification to adhere to the following term definitions as
consistently as possible.
The term noiseless according to the present invention refers to switching
between independent audio streams where packet sequence information is
preserved. The term synchronized header information refers to packets having
headers where packet sequence information is preserved. Packet sequence
information can include but is not limited to valid RTP information.
The term digital signal processor (DSP) includes but is not limited to a
device used to code or decode digitized voice samples according to a program or
application service.
The term digitized voice or voice includes but is not limited to audio byte
samples produced in a pulse code modulation (PCM) architecture by a standard
telephone circuit compressor/decompressor (CODEC).
The term packet processor refers to any type of packet processor that
creates packets for a packet-switched network. In one example, a packet
processor is a specialized microprocessor designed to examine and modify
Ethernet packets according to a program or application service.
The term packetized voice refers to digitized voice samples carried within
a packet.
The term real time protocol (RTP) stream of audio refers to the sequence
of RTP packets associated with one channel of packetized voice.
The term switched virtual circuit (SVC) refers to a temporary virtual
circuit that is set up and used only as long as data is being transmitted. Once the
communication between the two hosts is complete, the SVC disappears. In
contrast, a permanent virtual circuit (PVC) remains available at all times.
HI. Audio Networking Environment
The present invention can be used in any audio networking environment.
Such audio networking environments can include but are not limited to a wide
area and/or local area network environment. In example embodiments, the
present invention is incorporated within an audio networking environment as a
stand-alone unit or as part of a media server, packet router, packet switch or other
network component. For brevity, the present invention is described with respect
to embodiments incorporated in a media server.
Media servers deliver audio on network links over one or more circuit-
switched and/or packet-switched networks to local or remote clients. A client can
be any type of device that handles audio including but not limited to a telephone,
cellular phone, personal computer, personal data assistant (PDA), set-top box,
console, or audio player. FIG. 1 is a diagram of a media server 140 in an voice
over the Internet example environment according to the present invention. This
example includes a telephone client 105, public-switched telephone network
(PSTN) 110, Softswitch 120, gateway 130, media server 140, packet-switched
network(s) 150, and computer client 155. Telephone client 105 is any type of
phone (wired or wireless) that can send and receive audio over PSTN 110. PSTN
110 is any type of circuit-switched network(s). Computer client 155 can be a
personal computer.
Telephone client 105 is coupled through a public-switched telephone
network (PSTN) 110, gateway 130 and network 150 to media server 140. In this
example, call signaling and control is separated from the media paths or links that
carry audio. Softswitch 120 is provided between PSTN 110 and media server
140. Softswitch 120 supports call signaling and control to establish and remove
voice calls between telephone client 105 and media server 140. In one example,
Softswitch 120 follows the Session Initiation Protocol (SIP). Gateway 130 is
responsible for converting audio passing to and from PSTN 110 and network 150.
This can include a variety of well-known functions such as translating a circuit-
switched telephone number to an Internet Protocol (IP) address and vice versa.
Computer client 155 is coupled over network 150 to media server 140.
A media gateway controller (not shown) can also use SIP to support call signaling
and control to establish and breakdown links such as voice calls between
computer client 155 and media server 140. An application server(not shown) can
also be coupled to media server 140 to support VOL? services and applications.
The present invention is described in terms of these example
environments. Description in these terms is provided for convenience only. It is
not intended that the invention be limited to application in these example
environments involving a media server, router, switch, network component, or
stand-alone unit within a network. In fact, after reading the following description,
it will become apparent to a person skilled in the relevant art how to implement
the invention in alternative environments known now or developed in the future.
IV. Media Server, Services and Resources
FIG. 2 is a diagram of an example media platform 200 according to one
embodiment the present invention. Platform 200 provides scalable VOIP
telephony. Media platform 200 includes a media server 202 coupled to
resource(s) 210, media service(s) 212, and interface(s) 208. Media server 202
includes one or more applications 210, a resource manager 220 and audio
processing platform 230. Media server 202 provides resources 210 and services
212. Resources 210 include, but are not limited to modules 21 la-f, as shown in
FIG 2. Resource modules 21 la-f include conventional resources such as play
announcements/collect digits IVR resources 211a, tone/digit voice scanning
resource 21 lb, transcoding resource 21 lc, audio record/play resource 21 Id, text-
to-speech resource 21 le, and speech recognition resource 21 If. Media services
212 include, but are not limited to, modules 213a-e, as shown in FIG. 2. Media
services modules 213a-e include conventional services such as telebrowsing
213a, voice mail service 213b, conference bridge service 213c, video streaming
213d, and a VOIP gateway 213e.
Media server 202 includes an application central processing unit (CPU)
210, a resource manager CPU 220, and an audio processing platform 230.
Application CPU 210 is any processor that supports and executes program
interfaces for applications and applets. Application CPU 210 enables platform
200 to provide one or more of the media services 212. Resource manager CPU
220 is any processor that controls connectivity between resources 210 and the
application CPU 210 and/or audio processing platform 230. Audio processing
platform 230 provides communications connectivity with one or more of the
network interfaces 208. Media platform 200 through audio processing platform
230 receives and transmits information via network interface 208. Interface 208
can include, but it not limited to, Asynchronous Transfer Mode (ATM) 209a,
local area network (LAN) Ethernet 209b, digital subscriber line (DSL) 209c,
cable modem 209d, and channelized T1-T3 lines 209e.
V. Audio Processing Platform with a Packet/Cell Switch for Noiseless
Switching of Independent Audio Streams
In one embodiment of the present invention, audio processing platform
230 includes a dynamic fully-meshed cell switch 304 and other components for
the reception and processing of packets, such as Internet Protocol (IP) packets.
Platform 230 is shown in FIG. 3A with regard to audio processing including
noiseless switching according to the present invention.
As illustrated, audio processing platform 230 includes a call control and
audio feature manager 302, cell switch 304 (also referred to as a packet/cell
switch to indicate cell switch 304 can be a cell switch or packet switch), network
connections 305, network interface controller 306, and audio channel processors
308. Network interface controller 306 further includes packet processors 307.
Call control and audio feature manager 302 is coupled to cell switch 304, network
interface controller 306, and audio channels processors 308. In one
configuration, call control and audio feature manager 302 is connected directly
to the network interface controller 306. Network interface controller 306 then
controls packet processor 307 operation based on the control commands sent by
call control and audio feature manager 302.
In one embodiment, call control and audio feature manager 302 controls
cell switch 304, network interface controller 306 (including packet processors
307), and audio channel processors 308 to provide noiseless switching of
independent audio streams according to the present invention. This noiseless
switching is described further below with respect to FIGs. 6-9. An embodiment
of the call control and audio feature manager 302 according to the present
invention is described further below with respect to FIG. 3B.
Network connections 305 are coupled to packet processors 307. Packet
processors 307 are also coupled to cell switch 304. Cell switch 304 is coupled
in tum to audio channel processors 308. In one embodiment, audio channel
processors 308 include four channels capable of handling four calls, i.e., there are
four audio processing sections. In alternative embodiments, there are more or
less audio channel processors 308.
Data packets, such as IP packets, that include payloads having audio data
arrive at network connections 305. In one embodiment, packet processors 307
comprise one or more or eight 100Base-TX full-duplex Ethernet links capable of
high speed network traffic in the realm of 300,000 packets per second per link.
In another embodiment, packet processors 307 are capable of 1,000 G.711 voice
ports per link and/or 8,000 G.711 voice channels per system.
In additional embodiments, packet processors 307 recognize the IP
headers of packets and handle all RTP routing decisions with a minimum of
packet delay or jitter.
In one embodiment of the present invention, packet/cell switch 304 is a
non-blocking switch with 2.5Gbps of total bandwidth. In another embodiment,
the packet/cell switch 204 has 5Gbps of total bandwidth.
In one embodiment, the audio channel processors 308 comprise any audio
source, such as digital signal processors, as described in further detail with
regards to FIG. 4. The audio channel processors 308 can perform audio related
services including one or more of the services 211a-f.
VI. Example Audio Processing Platform Implementation
FIG. 4 shows one example implementation which is illustrative and not
intended to limit the present invention. As shown in FIG. 4, audio processing
platform 230 can be a shelf controller card (SCC). System 400 embodies one
such SCC. System 400 includes cell switch 304, call control and audio feature
manager 302, a network interface controller 306, interface circuitry 410, and
audio channel processors 308a-d.
More specifically,, system 400 receives packets at network connections
424 and 426. Network connections 424 and 426 are coupled to network interface
controller 306. Network interface controller 306 includes packet processors
307a-b. Packet processors 307a-b comprise controllers 420, 422, forwarding
tables 412,416, and forwarding processor (EPIF) 414,418. As shown in FIG. 4,
packet processor 307a is coupled to network connection 424. Network
connection 424 is coupled to controller 420. Controller 420 is coupled to both
forwarding table 412 andEPIF414. Packet processor 307b is coupled to network
connection 426. Network connection 426 is coupled to controller 422.
Controller 422 is coupled to both forwarding table 416 and EPIF 418.
In one embodiment, packet processors 307 can be implemented on one or
more LAN daughtercard modules. In another embodiment, each network
connection 424 and 426 can be a 100Base-TX or 1000Base-T link.
The IP packets received by the packet processors 307 are processed into
internal packets. When a cell layer is used, the internal packets are then
converted to cells (such as ATM cells' by a conventional segmentation and
reassembly (SAR) module). The cells are forwarded by packet processors 307
to cell switch 304. The packet processors 307 are coupled to the cell switch 304
via cell buses 428,430,432,434. Cell switch 304 forwards the cells to interface
circuitry 410 via cell buses 454,456,458,460. Cell switch 304 analyzes each of
the cells and forwards each of the cells to the proper cell bus of cell buses 454,
456, 458, 460 based on an audio channel for which that cell is destined. Cell
switch 304 is a dynamic, fully-meshed switch.
In one embodiment, interface circuitry 410 is a backplane connector.
The resources and services available for the processing and switching of
the packets and cells in system 400 are provided by call control and audio feature
manager 304. Call control and audio feature manager 302 is coupled to cell
switch 402 via a processor interface (PDF) 436, a SAR, and a local bus 437. Local
bus 437 is further coupled to a buffer 438. Buffer 438 stores and queues
instructions between the call control and'audio feature manager 302 and the cell
switch 304.
Call control and audio feature manager 302 is also coupled to a memory
module 442 and a configuration module 440 via bus connection 444. In one
embodiment, configuration module 440 provides control logic for the boot-up,
initial diagnostic, and operational parameters of call control and audio feature
manager 302. In one embodiment, memory module 442 comprises dual in-line
memory modules (DIMMs) for random access memory (RAM) operations of call
control and audio feature manager 302.
Call control and audio feature manager 302 is further coupled to interface
circuitry410. A network conduit 408 couples resource manager CPU 220 and/or
application CPU 210 to the interface circuitry 410. In one embodiment, call
control and audio feature manager 302 monitors the status of the interface
circuitry 410 and additional components coupled to the interface circuitry 410.
In another embodiment, call control and audio feature manager 302 controls the
operations of the components coupled to the interface circuitry 410 in order to
provide the resources 210 and services 212 of platform 200.
A console port 470 is also coupled to call control and audio feature
manager 302. Console port 470 provides direct access to the operations of call
control and audio feature manager 302. For example, one could administer the
operations, re-boot the media processor, or otherwise affect the performance of
call control and audio feature manager 302 and thus the system 400 using the
console port 470.
Reference clock 468 is coupled to interface circuitry 410 and other
components of the system 400 to provide consistent means of time-stamping the
packets, cells and instructions of the system 400.
Interface circuitry 410 is coupled to each of audio channel processors
308a-308d. Each of the processors 308 comprise a PIF 476, a group 478 of one
or more card processors (also referred to as "bank" processors), and a group 480
of one or more digital signal processors (DSP) and SDRAM buffers. In one
embodiment, there are four card processors in: group 478 and 32 DSPs in group
480. In such an embodiment, each card processor of group 478 would access and
operate with eight DSPs of group 480.
VII. Call Control and Audio Feature Manager
FIG. 3B is a block diagram of call control and audio feature manager 302
according to one embodiment of the present invention. Call control and audio
feature manager 302 is illustrated functionally as processor 302. Processor 302
comprises a call signaling manager 352, system manager 354, connection
manager 356, and feature controller 358.
Call signaling manager 352 manages call signaling operation such as call
establishment and removal, interface with a Softswitch, and handling signaling
protocols like SIP.
System manager 354 performs bootstrap and diagnostic operations on the
components of system 230. System manager 354 further monitors the system 230
and controls various hot-swapping and redundant operation.
Connection manager 356 manages EPIF forwarding tables, such as tables
412 and 416, and provides the routing protocols (such as Routing Information
Protocol (RIP), Open Shortest Path First (OSPF), and the like). Further, the
connection manager 356 establishes internal ATM permanent virtual circuits
(PVC) and/or SVC. In one embodiment, the connection manager 356 establishes
bi-directional connections between the network connections, such as network
connections 424 and 426, and the DSP channels, such as DSPs 480a-d, so that
data flows can be sources or processed by a DSP or other type of channel
processor.
In another embodiment, connection manager 356 abstracts the details of
the EPIF and ATM hardware. Call signaling manager 352 and the resource
manager CPU 220 can access these details so that their operations are based on
the proper service set and performance parameters.
Feature controller 358 provides communication interfaces and protocols
such as, H.323, and MGCP (Media Gateway Control Protocol).
In one embodiment, card processors 478a-d function as controllers with
local managers for the handling of instructions from the call control and audio
feature manager 302 and any of its modules: call signaling manager 352, system
manager 354, connection manager 356, and feature controller 358. Card
processors 478a-d then manage the DSP banks, network interfaces and media
streams, such as audio streams.
In one embodiment, the DSPs 480a-d provide the resources 210 and
services 212 of platform 200.
In one embodiment, call control and audio feature manager 302 of the
present invention exercises control over the EPIFof the present invention through
the use of applets. In such an embodiment, the commands for configuring
parameters (such as port MAC address, port IP address, and the like), search table
management, statistics uploading, and the like, are indirectly issued through
applets.
The EPIF provides a search engine to handle the functionality related to
creating, deleting and searching entries. Since the platform 200 operates on the
source and destination of packets, the EPIF provides search functionality of
sources and destinations. The sources and destinations of packets are stored in
search tables for incoming (ingress) and outgoing (egress) addresses. The EPIF
can also manage RTP header information and evaluating relative priorities of
egress audio streams to be transmitted as described in further detail below.
VIII. Audio Processing Platform Operation
The operation of audio processing platform 230 is illustrated in the flow
diagrams of FIGs. 5A and 5B. FIG. 5A is a flow diagram showing the
establishment of a call and ingress packet processing according to an embodiment
of the present invention. FIG. 5B is a flow diagram showing egress packet
processing and call completion according to an embodiment of the present
invention.
A. Ingress Audio Streams
In FIG. 5A, the process for an ingress (also called inbound) audio stream
starts at step 502 and immediately proceeds to step 504.
In step 504, call control and audio feature manager 302 establishes a call
with a client communicating via the network connections 305. In one
embodiment, call control and audio feature manager 302 negotiates and
authorizes access to the client. Once client access is authorized, call control and
audio feature manager 302 provides IP and UDP address information for the call
to the client. Once the call is established, the process immediately proceeds to
step 506.
In step 506, packet processors 307 receive IP packets carrying audio via
the network connections 305. Any type of packet can be used including but not
limited to IP packets, such as Appletalk, IPX, or other type of Ethernet packets.
Once a packet is received, the process proceeds to step 508.
In step 508, packet processors 307 check IP and UDP header address in
search table to find associated SVC, and then convert the VOIP packets into
interna] packets. Such internal packets for example can be made up of a payload
and control header as described further below with respect to FIG. 7B. Packet
processors 307 then construct packets using at least some of the data and routing
information and assign a switched virtual circuit (SVC). The SVC is associated
with one of the audio channel processors 308, and in particular with one of
respective DSP that will process the audio payload.
When a cell layer is used, internal packets are further converted or merged
into cells, such as ATM cells. In this way, audio payloads in the internal packets
are converted to audio payloads in a stream of one or more ATM cells. A
conventional segmentation and reassembly (SAR) module can be used to convert
internal packets to ATM cells. Once the packets are converted into the cells, the
process proceeds to step 510.
In step 510, cell switch 304 switches the cells to the proper audio channel
of the audio channel processors 308 based on the SVC. The process proceeds to
step 512.
In step 512, audio channel processors 308 convert the cells into packets.
Audio payloads in the arriving ATM cells for each channel are converted to audio
payloads in a stream of one or more packets. A conventional SAR module can
be used to convert ATM to packets. Packets can be internal egress packets or IP
packets with audio payloads. Once the cells are converted into the internal
packets, the process proceeds to step 514.
In step 514, audio channel processors 308 process the audio data of the
packets in the respective audio channels. In one embodiment, the audio channels
are related to one or more of the media services 213a-e. For example, these
media services can be telebrowsing, voice mail, conference bridging (also called
conference calling), video streaming, VOIP gateway services, telephony, or any
other media service for audio content.
B. Egress Audio Streams
In FIG. 5B, the process for an egress (also called outbound) audio stream
starts at step 522 and immediately proceeds to step 524.
In step 524, call control and audio feature manager 302 identifies an
audio source for noiseless switch over. This audio source can be associated with
an established call or other media service. Once the audio source is identified, the
process immediately proceeds to step 526.
In step 526, an audio source creates packets. In one embodiment, a DSP
in audio channel processor 308 is an audio source. Audio data can be stored in
a SDRAM associated with the DSP. This audio data is then packetized by a DSP
into packets. Any type of packet can be used including but not limited to internal
packets or IP packets, such as Ethernet packets. In one preferred embodiment, the
packets are internal egress packets generated as described with respect to FIG.
7B.
In step 528, an audio channel processor 308 converts the packets into
cells, such as ATM cells. Audio payloads in the packets are converted to audio
payloads in a stream of one or more ATM cells. In brief, the packets are parsed
and the data and routing information analyzed. Audio channel processor 308 then
construct cells using at least some of the data and routing information and assigns
a switched virtual circuit (SVC). A conventional SAR module can be used to
convert packets to ATM cells. The SVC is associated with one of the audio
channel processors 308, and in particular with a circuit connecting the respective
DSP of the audio source and a destination port 305 of NIC 306. Once the packets
are converted into the cells, the process proceeds to step 530.
In step 530, cell switch 304 switches the cells of an audio channel of the
audio channel processors 308 to a destination network connection 305 based on
the SVC. The process proceeds to step 532.
In step 532, packet processors 307 convert the cells into IP packets.
Audio payloads in the arriving ATM cells for each channel are converted to audio
payloads in a stream of one or more internal packets. A conventional SAR
module can be used to convert ATM to internal packets. Any type of packet can
be used including but not limited to IP packets, such as Ethernet packets. Once
the cells are converted into the packets, the process proceeds to step 534.
In step 534, each packet processor 307 further adds RTP, IP, and UDP
header information. A search table is checked to find IP and UDP header address
information associated with the SVC. IP packets are then sent carrying audio via
the network connections 305 over a network to a destination device (phone,
computer, palm device, PDA, etc.). Packet processors 307 process the audio data
of the packets in the respective audio channels. In one embodiment, the audio
channels are related to one or more of the media services 213a-e. For example,
these media services can be telebrowsing, voice mail, conference bridging (also
called conference calling), video streaming, VOIP gateway services, telephony,
or any other media service for audio content.
IX. Noiseless Switching of Egress Audio Streams
According to the one aspect of the present invention, audio processing
platform 230 noiselessly switches between independent egress audio streams.
Audio processing platform 230 is illustrative. The present invention as it relates
to noiseless switching of egress audio stream can be used in any media server,
router, switch, or audio processor and is not intended to be limited to audio
processing platform 230.
A. Cell Switch - Internal Audio Sources
FIG. 6A is diagram of a noiseless switch over system that carries out cell
switching of independent egress audio streams generated by internal audio
sources according to an embodiment of the present invention. FIG. 6A shows an
embodiment of a system 600A for egress audio stream switching from internal
audio sources. System 600A includes components of audio processing platform
230 configured for an egress audio stream switching mode of operation. In
particular, as shown in FIG. 6A, system 600A includes call control and audio
feature controller 302 coupled to a number n of internal audio sources 604n, cell
switch 304, and network interface controller 306. Internal audio sources 604a-
604n can be two or more audio sources. Any type of audio source can be used
including but not limited to DSPs. In one example, DSPs 480 can be audio
sources. To generate audio, audio sources 604 can either create audio internally
and/or convert audio received from external sources.
Call control and audio feature controller 302 further includes an egress
audio controller 610. Egress audio controller 610 is control logic that issues
control signals to audio sources 604n, cell switch 304, and/or network interface
controller 306 to carry out noiseless switching between independent egress audio
streams according to the present invention. The control logic can implemented
in software, firmware, microcode, hardware or any combination thereof.
A cell layer including SARs 630, 632,634 is also provided. SARs 630,
632 are coupled between cell switch 304 and each audio source 604a-n. SAR 634
is coupled between cell switch 304 and NIC 306.
In one embodiment, independent egress audio streams involve streams of
IP packets with RTP information and internal egress packets. Accordingly, it is
helpful to first describe IP packets and internal egress packets (FIGs. 7A-7B).
Next, system 600A and its operation is described in detail with respect to
independent egress audio streams (FIGs. 8-9).
B. Packets
In one embodiment, the present invention uses two types of packets: (1)
IP packets with RTP information and (2) internal egress packets. Both of these
types of packets are shown and described with respect to examples in FIGs. 7A
and 7B. IP packets 700A are sent and received over a external packet-switched
network by packet processors 307 in NIC 306. Internal egress packets 700B are
generated by audio sources (e.g. DSPs) 604a-604n.
1. IP Packets with RTP Information
A standard Internet Protocol (IP) packet 700A is shown in FIG. 7A. IP
packet 700A is shown with various components: media access control (MAC)
field 704, IP field 706, user datagram protocol (UDP) field 708, RTP field 710,
payload 712 containing digital data, and cyclic redundancy check (CRC) field
714. Real-Time Transport Protocol (RTP) is a standardized protocol for carrying
periodic data, such as digitized audio, from a source device to a destination
device. A companion protocol, Real-Time Control Protocol (RTCP), can also be
used with RTP to provide information on the quality of a session.
More specifically, the MAC 704 and IP 706 fields contain addressing
information to allow each packet to traverse an IP network interconnecting two
devices (origin and destination). UDP field 708 contains a 2-byte port number
that identifies a RTP/audio stream channel number so that it can be internally
routed to the audio processor destination when received from the network
interface. In one embodiment of the present invention, the audio processor is a
DSP, as described herein.
RTP field 710 contains a packet sequence number and timestamp.
Payload 712 contains the digitized audio byte samples and can be decoded by the
endpoint audio processors. Any payload type and encoding scheme for audio
and/or video types of media compatible with RTP can be used as would be
apparent to a person skilled in the art given this description. CRC field 714
provides a way to verify the integrity of the entire packet. See, the description of
RTP packets and payload types described by D. Collins, Carrier Grade Voice
over IP, pp. 52-72 (the text of the entire book of which is incorporated herein by
reference).
2. Internal Egress Packets
FIG. 7B illustrates an example internal egress packet of the present
invention in greater detail. Packet 700B includes a control (CTRL) header 720
and a payload 722. The advantage of internal egress packet 700B is it is simpler
to create and smaller in size than IP packet 700A. This reduces the burden and
work required of audio sources and other components handling the internal egress
packets.
In one embodiment, audio sources 604a-604n are DSPs. Each DSP adds
a CTRL header 720 in front of a payload 722 that it creates in for a respective
audio stream. CTRL 720 is then used to relay control information downstream.
This control information for example can be priority information associated with
a particular egress audio stream.
Packet 700B is converted to one or more cells, such as ATM cells, and
sent internally over cell switch 304 to a packet processor 307 in network interface
controller 306. After the cells are converted to internal egress packets, packet
processor 307 decodes and removes internal header CTRL 720. The rest of the
IP packet information is added before the payload 722 is transmitted as an IP
packet 700A onto an IP network. This achieves an advantage as processing work
at the DSPs is reduced. DSPs only have to add a relatively short control header
to payloads. The remaining processing work of adding information to create
valid IP packets with RTP header information can be distributed to packet
processors) 307.
C. Priority Levels
Network interface controller (NIC) 306 processes all internal egress
packets, as well as all egress IP packets destined for the external network. Thus,
NIC 306 can make final forwarding decisions about each packet sent to it based
on the content of each packet. In some embodiments, NIC 306 manages the
forwarding of egress IP packets based on priority information. This can include
switching over to an audio stream of egress IP packets with a higher priority and
buffering or not forwarding another audio stream of egress IP packets with a
lower priority.
In one embodiment, internal audio sources 604a-604n determine priority
levels. Alternatively, NIC 306 can determine a priority for audio received from
an external source at NIC 306. Any number of priority levels can be used. The
priority levels distinguish the relative priority of audio sources and their
respective audio streams. Priority levels can be based on any criteria selected by
a user including, but not limited to, time of day, identity or group of the caller or
callee, or other similar factors relevant to audio processing and media services.
Components of the system 600 filter and forward the priority level information
within the audio stream. In one embodiment, a resource manager in system 600
can interact with external systems to alter the priority levels of audio streams. For
example, an external system can be an operator informing the system to queue a
billing notice or advertisement on a call. Thus, the resource manager is capable
of barging into audio streams. This noiseless switch over can be triggered by user
or automatically based on certain predefined events such as signaling conditions
like on-hold condition, emergency event, or timed event.
D. Noiseless Fully Meshed Cell Switch
System 600A can be thought of as a "free pool" of multiple input (ingress)
and output (egress) audio channels because a fully meshed packet/cell switch 304
is used to switch egress audio channels to participate in any given call. Any
egress audio channel can be called upon to participate in a telephone call at any
time. During both the initial call setup and while the call is in session, any egress
audio channel can be switched into and out of the call. The fully meshed
switching capability of system 600A of the present invention provides a precise
noiseless switching functionality which does not drop or corrupt the IP packets
or the cells of the present invention. In addition, a two-stage egress switching
technique is used.
E. Two-Stage Egress Switching
System 600A includes at least two stages of switching. In terms of egress
switching, the first stage is cell switch 304. The first stage is cell-based and uses
switched virtual circuits (SVCs) to switch audio streams from separate physical
sources (audio sources 604a-604n) to a single destination egress network
interface controller (NIC 306). Priority information is provided in the CTRL
header 720 of cells generated by the audio sources. The second stage is contained
within the egress NIC 306 such that it selects which of the audio streams from
multiple audio sources (604a-604n) to process and send over a packet network
such as an packet-switched IP network. This selection of which audio streams to
forward can be performed by NIC 306 is based on the priority information
provided in the CTRL headers 720. In this way, a second audio stream with a
higher priori ty can be forwarded by NIC 306 on the same channel as a first audio
stream. From the perspective of the destination device receiving the audio
streams, the insertion of the second audio stream on the channel is received as a
noiseless switch between independent audio streams.
More specifically, in one embodiment, the egress audio switching can
occur in a telephone call. A call is first established using audio source 604a by
negotiating with the destination device's MAC, IP, and UDP information, as
previously described. First audio source 604a begins generating a first audio
stream during the call. The first audio stream is made up of internal egress
packets having audio payload and CTRL header 720 information as described
with respect to packet format 700B. Internal egress packets egress on the channel
established for the call. Any type of audio payload including voice, music, tones,
or other audio data can be used. SAR 630 converts the internal packets to cells
for transport through cell switch 304 to SAR 634. SAR 634 then converts cells
back to internal egress packets prior to delivery to NIC 306.
During the flow from the audio source 604a, NIC 306 is decoding and
removing the CTRL header 720 and adding the appropriate RTP, UDP, IP, MAC,
and CRC fields, as previously described. CTRL header 720 includes the priority
field used by NIC 306 to process the packet and send a corresponding RTP
packet. NIC 306 evaluates the priority field. Given the relatively high priority
field (the first audio source 604a is the only transmitting source), NIC 306
forwards IP packets with synchronized RTP header information which carry the
first audio stream over the network to the destination device associated with the
call. (Note CTRL header 720 can also include RTP or other synchronized header
information which can be used or ignored by NIC 306 if NIC 306 generates and
adds RTP header information).
When the egress audio controller 610 determines a call event where a
noiseless switch over is to occur, a second audio source 604n begins generating
a second audio stream. Audio can be generated by audio source 604n directly or
by converting audio originally generated by external devices. The second audio
stream is made up of internal egress packets having audio payload and CTRL
header 720 information as described with respect to packet format 700B. Any
type of audio payload including voice, music, or other audio data can be used.
Assume the second audio stream is given a higher priority field than the first
audio stream. For example, the second audio stream can represent an
advertisement, emergency public service message, or other audio data that is
desired to have noiselessly inserted into the first channel established with the
destination device.
The second audio stream's internal egress packets are then converted to
cells by SAR 632. Cell switch 304 switches the cells to an SVC destined for the
same destination NIC 306 as the first audio stream. SAR 634 converts the cells
back to internal packets. NIC 306 now receives the internal packets for the first
and second audio streams. NIC 306 evaluates the priority field in each stream.
The second audio stream having internal packets with the higher priority are
converted to IP packets with synchronized RTP header information and
forwarded to the destination device. The first audio stream having internal
packets with the lower priority are either stored in a buffer or converted to IP
packets with synchronized RTP header information and stored in buffer. NIC 306
can resume forwarding the first audio stream when the second audio stream is
completed, after a predetermined time elapses, or when a manual or automatic
control signal is received to resume.
F. Call Event Triggering Noiseless Switch Over
The functionality of the priority field in an embodiment of noiseless
switching according to the present invention is now described with regard to
FIGs. 8, 9A and 9B.
In FIG. 8, a flow diagram of a noiseless switching routine 800 according
to one embodiment of the present invention is shown. For brevity, the noiseless
switching routine 800 is described with respect system 600.
Flow 800 begins at step 802 and proceeds immediately to step 804.
In step 804, call control and audio feature manager 302 establishes a call
from a first audio source 604a to a destination device. Call control and audio
feature manager 302 negotiates with the destination device to determine the
MAC, IP and UDP port to use in a first audio stream of IP packets sent over a
network.
Audio source 604a delivers a first audio stream on one channel for the
established call. In one embodiment, a DSP delivers the first audio stream of
internal egress packets on one channel to cell switch 304 and then to NIC 306.
The process proceeds to step 806.
In step 806, egress audio controller 610 sets a priority field for the first
audio source. In one embodiment, egress audio controller 610 sets the priority
field to a value of one. In another embodiment, the priority field is stored in the
CTRL header of the internally routed internal egress packets. The process
immediately proceeds to step 808.
In step 808, egress audio controller 610 determines the call's status In
one embodiment, egress audio controller 610 determines whether or not the call
allows or has been configured to allow call events to interact with it. In one
embodiment of the present invention, a call can be configured so that only
emergency call events will interrupt it. In another embodiment, a call can be
configured to receive certain call events based on either the caller(s) or callee(s)
(i.e., the one or more of the parties on the call). The process immediately
proceeds to step 810.
In step 810, egress audio controller 610 monitors for call events. In one
embodiment, a call event can be generated within the system 600, such as
notifications of time, weather, advertisements, billing ("please insert another
coin" or "you have 5 minutes remaining"). In another embodiment, call events
can be sent to the system 600, such as requests for news, sporting information,
etc. Egress audio controller 610 can monitor both internally and externally for
call events. The process proceeds immediately to step 812.
In step 812, egress audio controller 610 receives a call event. If not, then
egress audio controller 610 continues to monitor as stated in step 810. If so, then
the process proceeds immediately to step 814.
In step 814, egress audio controller 610 determines the call event and
performs the operations necessitated by the call event. The process then proceeds
to step 816 where it either ends or returns to step 802. In one embodiment, the
process 800 repeats for as long as the call continues.
In FIG. 9A-9C, flow diagram 900 of the call event processing for audio
stream switching based on priority according to one embodiment of the present
invention are shown. In one embodiment, flow 900 shows in more detail the
operations performed in step 814 of FIG. 8.
Process 900 starts at step 902 and proceeds immediately to step 904.
In step 904, egress audio controller 610 reads a call event for an
established call. In this operation, a first audio stream from source 604a is
already being sent from NIC 306 to a destination device as part of the established
call. The process proceeds to step 906.
In step 906, egress audio controller 610 determines whether the call event
includes a second audio source. If so, then the process proceeds to step 908. If
not, then the process proceeds to step 930.
In step 908, egress audio controller 610 determines the priority of the
second audio source. In one embodiment, egress audio controller 610 issues a
command to second audio source 604n that instructs the second audio source to
generate a second audio stream of internal egress packets. Priority information
for the second audio stream can be automatically generated by the second audio
source 604n or generated based on a command from the egress audio controller
610. The process then proceeds to step 910.
In step 910, a second audio source 604n begins generating a second audio
stream. The second audio stream is made up of internal egress packets having
audio payload and CTRL header 720 information as described with respect to
packet format 700B. Any type of audio payload including voice, music, or other
audio data can be used. Audio payload is meant broadly to also include audio
data included as part of video data. The process then proceeds to step 912.
In step 912, the second audio stream's egress packets are then converted
to cells. In one example, the cells are ATM cells. The process then proceeds to
step 914.
In step 914, cell switch 304 switches the cells to an SVC destined for the
same destination NIC 306 on the same egress channel as the first audio stream.
The process then proceeds to step 915.
As shown in step 915 of FIG. 9B, SAR 634 now receives cells for the first
and second audio streams. The cells are converted back to streams of internal
egress packets and have control headers that include the respective priority
information for the two audio streams.
In step 916, NIC 306 compares the priorities of the two audio streams. If
the second audio stream has a higher priority then the process proceeds to step
918. If not, then the process proceeds to step 930.
In step 918, the transmission of the first audio stream is held. For
example, NIC 306 buffers the first audio stream or even issues a control
command to audio source 604a to hold the transmission of the first audio source.
The process proceeds immediately to step 920.
In step 920, the transmission of the second audio stream starts. NIC 306
instructs packet processor(s) 307 to create IP packets having the audio payload
of the internal egress packets of the second audio stream. Packet processor(s) 307
add additional synchronized RTP header information (RTP packet information)
and other header information (MAC, IP, UDP fields) to the audio payload of the
internal egress packets of the second audio stream.
NIC 306 then sends the IP packets with synchronized RTP header
information on the same egress channel of the first audio stream. In this way, a
destination device receives the second audio stream noise instead of the first
audio stream. Moreover, from the perspective of the destination device this
second audio stream is received in real-time noiselessly without delay or
interruption. Steps 918 and 920 of course can be performed at the same time or
in any order. The process proceeds immediately to step 922.
As shown in FIG. 9C, NIC 306 monitors for the end of the second audio
stream (step 922). The process proceeds immediately to step 924.
In step 924, NIC 306 determines whether the second audio stream has
ended. In one example, NIC 306 reads a last packet of the second audio stream
which has a priority level lower than preceding packets. If so, then the process
proceeds immediately to step 930. If not, then the process proceeds to step 922.
In step 930, NIC 306 either continues to forward the first audio stream
(after step 906) or returns to forwarding the first audio stream (after steps 916 or
924). The process proceeds to step 932.
In one embodiment, NIC 306 maintains a priority level threshold value.
NIC 306 then increments and sets the threshold based on priority information in
the audio streams. When faced with multiple audio streams, NIC 306 forwards
the audio stream having priority information equal to or greater than the priority
level threshold value. For example, if the first audio stream had a priority value
of 1 then the priority level threshold value is set to 1 and the first audio stream is
transmitted (prior to step 904). When a second audio stream with a higher
priority is received at NIC 306, then NIC 306 increments the priority threshold
value to 2. The second audio stream is then transmitted as described above in
step 920. When the last packet of the second audio stream having a priority field
value set to 0 (or null or other special value) is read, then the priority level
threshold value is decremented back to 1 as part of step 924. In this case, the first
audio stream with priority information 1 is then be sent by NIC 306 as described
above with respect to step 930.
In step 932, egress audio controller 610 processes any remaining call
events. The process then proceeds to step 934 where it terminates until re-
instantiated. In one embodiment, the steps of the above-described process occur
substantially at the same time, such that the process can be run in parallel or in
an overlapping manner on one or more processors in the system 600.
G. Audio Data Flow
FIG. 6B is a diagram of audio data flow 615 in the noiseless switch over
system of FIG. 6A in one embodiment. In particular, FIG. 6B shows the flow of
internal packets from audio sources 604a-n to S ARs 630,632, the flow of cells
through cell switch 304 to SAR 634, the flow of internal packets between SAR
634 and packet processors 307, and the flow of IP packets from NIC 306 over the
network.
H. Other Embodiments
The present invention is not limited to internal audio sources or a cell
layer. Noiseless switch over can also be carried out in different embodiments
using internal audio sources only, internal and external audio sources, external
audio sources only, a cell switch or a packet switch. For example, FIG. 6C is
diagram of a noiseless switch over system 600C that carries out cell switching
between independent egress audio streams generated by internal audio source
604a-n and/or external audio sources (not shown) according to an embodiment
of the present invention. Noiseless switch over system 600C operates similar to
system 600A described in detail above except that noiseless switch over is made
to audio received from an external audio source. The audio is received in IP
packets and buffered at NIC 306 as shown in FIG. 6C. NIC 306 strips IP
information (stores it in forward table entry associated with external audio source
and destination device) and generates internal packets assigned to a SVC. SAR
634 converts the internal packets to cells and routes cells on the SVC on link 662
through switch 304 back through link 664 to SAR 634 for conversion to internal
packets. As described above, the internal packets are then processed by packet
processor 307 to create IP packets with synchronized header information. NIC
306 then sends the IP packets to destination device. In this way, a user at the
destination device is noiselessly switched over to receive audio from an external
audio source. FIG. 6D is diagram of audio data flow 625 for an egress audio
stream received from the external audio source in the noiseless switch over
system of FIG. 6C. In particular, FIG. 6D shows the flow of IP packets from an
external audio source (not shown) to NIC 306, the flow of internal packets from
NIC 306 to SAR 634, the flow of cells through cell switch 304 back to SAR 634,
the flow of internal packets between SAR 634 and packet processors 307, and the
flow of IP packets from NIC 306 over the network to a destination device (not
shown).
FIG. 6E is diagram of audio data flows 635,645 in a noiseless switch over
system 600E that carries out packet switching between independent egress audio
streams generated by internal and/or externa] audio sources according to an
embodiment of the present invention. Noiseless switch over system 600E
operates similar to systems 600A and 600 C described in detail above except that
a packet switch 694 is used instead of a cell switch 304. In this embodiment, a
cell layer including SARs 630, 632, 634 is omitted. In audio data flow 635,
internal packets flow through the packet switch 694 from internal audio sources
604a-n to packet processors 307. IP packets flow out to the network. In audio
data flow 645, IP packets from an external audio source (not shown) are received
at NIC 306. The audio is received in packets and buffered at NIC 306 as shown
in FIG. 6E. NIC 306 strips IP information (stores it in forward table entry
associated with external audio source and destination device) and generates
internal packets assigned to a SVC (or other type of path) associated with the
destination device. The internal packets are routed on the SVC through packet
switch 694 to NIC 306. As described above, the internal packets are then
processed by packet processor 307 to create IP packets with synchronized header
information. NIC 306 then sends the IP packets to destination device. In this
way, a user at the destination device is noiselessly switched over to receive audio
from an external audio source.
FIG. 6F is diagram of a noiseless switch over system 600F that carries out
switching between independent egress audio streams generated by only external
audio sources according to an embodiment of the present invention. No switch
or internal audio sources are required. NIC 306 strips IP information (stores it in
forward table entry associated with external audio source and destination device)
and generates internal packets assigned to a SVC (or other type of path)
associated with the destination device. The internal packets are routed on the
SVC to NIC 306. (NIC 306 can be a common source and destination point). As
described above, the internal packets are then processed by packet processor 307
to create IP packets with synchronized header information. NIC 306 then sends
the IP packets to destination device. In this way, a user at the destination device
is noiselessly switched over to receive audio from an external audio source.
Functionality described above with respect to the operation of egress
audio switching system 600 can be implemented in control logic. Such control
logic can be implemented in software, firmware, hardware or any combination
thereof.
X. Conference Call Processing
A. Distributed Conference Bridge
FIG. 10 is a diagram of a distributed conference bridge 1000 according to
one embodiment of the present invention. Distributed conference bridge 1000 is
coupled to a network 1005. Network 1005 can be any type of network or
combination of networks, such as, the Internet. For example, network 1005 can
include a packet-switched network or a packeted-switched network in
combination with a circuit-switched network. A number of conference call
participants Cl-CN can connect through network 1005 to distributed conference
bridge 1000. For example, conference call participants Cl-CN can place a VOIP
call through network 1005 to contact distributed conference bridge 1000.
Distributed conference bridge 1000 is scalable and can handle any number of
conference call participants. For example, distributed conference bridge 1000 can
handle conference calls between two conference call participants up to 1000 or
more conference call participants.
As shown in FIG. 10, distributed conference bridge 1000 includes a
conference call agent 1010, network interface controller (NIC) 1020,switch 1030,
and audio source 1040. Conference call agent 1010 is coupled to NIC 1020,
switch 1030 and audio source 1040. NIC 1020 is coupled between network 1005
and switch 1030. Switch 1030 is coupled between NIC 1020 and audio source
1040. A look-up table 1025 is coupled to NIC 1020. Look-up table 1025 (or a
separate look-up table not shown) can also be coupled to audio source 1040.
Switch 1030 includes a muiticaster 1050. NIC 1020 includes a packet processor
1070.
Conference call agent 1010 establishes a conference call for a number of
participants. During a conference call, packets carrying audio, such as digitized
voice, flow from the conference call participants Cl-CN to the conference bridge
1000. These packets can be IP packets including, but not limited to, RTP/RTCP
packets. NIC 1020 receives the packets and forwards the packets along links
1028 to switch 1030. Links 1028 can be any type of logical and/or physical links
such as PVCs or SVCs. In one embodiment, NIC 1020 converts IP packets (as
described above with respect to FIG. 7A) to internal packets which only have a
header and payload (as described with respect to FIG. 7B). The use of the
internal packets further reduces processing work at audio source 1040. Incoming
packets processed by NIC 1020 can also be combined by a SAR into cells, such
as ATM cells, and sent over link(s) 1028 to switch 1030. Switch 1030 passes the
incoming packets from NIC 1020 (or cells) to audio source J 040 on link(s) 1035.
Link(s) 1035 can also be any type of logical and/or physical link including, but
not limited to, a PVC or SVC.
Audio provided over links 1035 is referred to in this conference bridge
processing context as "external audio" since it originates from conference call
participants over network 1005. Audio can also be provided internally through
one or more links 1036 as shown in FIG. 10. Such "internal audio" can be
speech, music, advertisements, news, or other audio content to be mixed in the
conference call. The internal audio can be provided by any audio source or
accessed from a storage device coupled to conference bridge 1000.
Audio source 1040 mixes audio for the conference call. Audio source
1040 generates outbound packets containing the mixed audio and sends the
packets over link(s) 1045 to switch 1030. In particular, audio source 1040
generates a fully mixed audio stream of packets and a set of partially mixed audio
streams. In one embodiment, audio source 1040 (or "mixer" since it is mixing
audio) dynamically generates the appropriate fully mixed and partially mixed
audio streams of packets having conference identifier information (CID) and
mixed audio during the conference call. The audio source retrieves the
appropriate CED information of conference call participants from a relatively
static look-up table (such as table 1025 or a separate table closer to audio source
1040) generated and stored at the initiation of the conference call.
Multicaster 1050 multicasts the packets in the fully mixed audio stream
and a set of partially mixed audio streams. In one embodiment, multicaster 1050
replicates the packets in each of the fully mixed audio stream and set of partially
mixed audio streams N times which corresponds to the N number of conference
call participants. The N replicated packets are then sent to endpoints in NIC 1020
over the N switched virtual circuits (SVC1-SVCN), respectively. One advantage
of distributed conference bridge 1000 is that audio source 1040 (i.e., the mixing
device) is relieved of the work of replication. This replication work is distributed
to multicaster 1050 and switch 1030.
NIC 1020 then processes outbound packets arriving on each SVC1-SVCN
to determine whether to discard or forward the packets of the fully mixed and
partially mixed audio streams to a conference call participant Cl-CN. This
determination is made based on packet header information in real-time during a
conference call. For each packet arriving on a SVC, NIC 1020 determines based
on packet header information, such as TAS and IAS fields, whether the packet is
appropriate for sending to a participant associated with the SVC. If yes, then the
packet is forwarded for further packet processing. The packet is processed into
a network packet and forwarded to the participant. Otherwise, the packet is
discarded. In one embodiment, the network packet is an IP packet which includes
the destination call participant's network address information (IP/UDP address)
obtained from a look-up table 1025, RTP/RTCP packet header information (time
stamp/sequence information), and audio data. The audio data is the mixed audio
data appropriate for the particular conference call participant. The operation of
distributed conference bridge 1000 is described further below with respect to an
example look-up table 1025 shown in FIG. 11, flowchart diagrams shown in
FIGs. 12 and 13A-13C, and example packet diagrams shown in FIGs. 14A, 14B
and 15.
B. Distributed Conference Bridge Operation
FIG. 12 shows a routine 1200 for establishing conference bridge
processing according to the present invention. (Steps 1200-1280). In step 1220,
a conference call is initiated. A number of conference call participants Cl-CN
dial distributed conference bridge 1000. Each participant can use any VOIP
terminal including, but not limited to, a telephone, computer, PDA, set-top box,
network appliance, etc. Conference call agent 1010 performs conventional IVR
processing to acknowledge that a conference call participant wishes to participate
in a conference call and obtains the network address of each conference call
participant. For example, the network address information can include, but is not
limited to, IP and/or UDP address information.
In step 1240, look-up table 1025 is generated. Conference call agent 1010
can generate the look-up table or instruct NIC 1020 to generate the look-up table.
As shown in the example on FIG. 11, look-up table 1025 includes N entries
corresponding to the N conference call participants in the conference call initiated
in step 1220. Each entry in look-up table 1025 includes an SVC identifier,
conference ID (CID), and network address information. The SVC identifier is
any number or tag that identifies a particular SVC. In one example, the SVC
identifier is a Virtual Path Identifier and Virtual Channel Identifier (VPI/VCI).
Alternatively, the SVC identifier or tag information can be omitted from look-up
table 1025 and instead be inherently associated with the location of the entry in
the table. For example, a first SVC can be associated with the first entry in the
table, a second SVC can be associated with a second entry in the table, and so
forth. The CID is any number or tag assigned by conference call agent 1010 to
a conference call participant C1-CN. The network address information is the
network address information collected by conference call agent 1010 for each of
the N conference call participants.
In step 1260, NIC 1020 assigns respective SVCs to each of the
participants. For N conference call participants then N SVCs are assigned.
Conference call agent 1010 instructs NIC 1020 to assign N SVCs. NIC 1020 then
establishes N SVC connections between NIC 1020 and switch 1030. In step
1280, the conference call then begins. Conference call agent 1010 sends a signal
to NIC 1020 and switch 1030 and audio source 1040 to begin conference call
processing. Although FIG. 12 is described with respect to SVCs and SVC
identifiers, the present invention is not so limited and any type of link (physical
and/or logical) and link identifier can be used. Also, in embodiments where an
internal audio source is included, conference call agent 1010 adds the internal
audio source as one of the potential N audio participants whose input is to be
mixed at audio source 1040.
The operation of distributed conference bridge 1000 during conference
call processing is shown in FIGs. I3A-13C (steps 1300-1398). Control begins at
step 1300 and proceeds to step 1310. In step 1310, audio source 1040 monitors
energy in the incoming audio streams of the conference call participant CI-CN.
Audio source 1040 can be any type of audio source including, but not limited to,
a digital signal processor (DSP). Any conventional technique for monitoring the
energy of a digitized audio sample can be used. In step 1320, audio source 1040
determines a number of active speakers based on the energy monitored in step
1310. Any number of active speakers can be selected. In one embodiment, a
conference call is limited to three active speakers at a given time. In this case, up
to three active speakers are determined which correspond to the up to three audio
streams having the most energy during the monitoring in step 1320.
Next, audio source 1040 generates and sends fully mixed and partially
mixed audio streams (steps 1330-1360). In step 1330, one fully mixed audio
stream is generated. The fully mixed audio stream includes the audio content of
the active speakers determined in step 1320. In one embodiment, the fully mixed
audio stream is an audio stream of packets with packet headers and payloads.
Packet header information identifies the active speakers whose audio content is
included in the fully mixed audio stream. In one example, as shown in FIG. 14A
audio source 1040 generates an outbound internal packet 1400 having a packet
header 1401 with TAS. IAS, and Sequence fields and a payload 1403. The TAS
field lists CIDs of all of the current active speaker calls in the conference call.
The IAS field lists CIDs of the active speakers whose audio content is in the
mixed stream. The sequence information can be a timestamp, numeric sequence
value, or other type of sequence information. Other fields (not shown) can
include checksum or other packet information depending upon a particular
application. In the case of a fully mixed audio stream, the TAS and IAS fields are
identical. Payload 1403 contains a portion of the digitized mixed audio in the
fully mixed audio stream.
In step 1340, audio source 1040 sends the fully mixed audio stream
generated in step 1330 to switch 1030. Eventually, passive participants in the
conference call (that is those determined not to be in the number of active
speakers determined in step 1320), will hear mixed audio from the fully mixed
audio stream.
In step 1350, audio source 1040 generates a set of partially mixed audio
streams. The set of partially mixed audio streams is then sent to switch 1030
(step 1360). Each of the partially mixed audio streams generated in step 1350 and
sent in step 1360 includes the mixed audio content of the group of identified
active speakers determined in step 1320 minus the audio content of a respective
recipient active speaker. The recipient active speaker is the active speaker within
the group of active speakers determined in step 1320 towards which a partially
mixed audio stream is directed.
In one embodiment, audio source 1040 inserts in packet payloads the
digital audio from the group of identified active speakers minus the audio content
of the recipient active speaker. In this way, the recipient active speaker will not
receive audio corresponding to their own speech or audio input. However, the
recipient active speaker will hear the speech or audio input of the other active
speakers. In one embodiment, packet header information is included in each
partially mixed audio stream to identify active speakers whose audio content is
included in the respective partially mixed audio stream. In one example, audio
source 1040 uses the packet format of FIG.14A and inserts one or more
conference identification numbers (CIDs) into TAS and IAS header fields of
packets. The TAS field lists CIDs of all of the current active speakers in the
conference call. The IAS field lists CIDs of the active speakers whose audio
content is in the respective partially mixed stream. In the case of a partially
mixed audio stream, the TAS and IAS fields are not identical since the IAS field
has one less CID. In one example, to build packets in steps 1330 and 1350, audio
source 1040 retrieves the appropriate CID information of conference call
participants from a relatively static look-up table (such as table 1025 or a separate
table) generated and stored at the initiation of the conference call.
For example, in a conference call where there are 64 participants (N = 64)
of which three are identified as active speakers (1-3), then one fully mixed audio
stream will contain audio from all three active speakers. This fully mixed stream
is eventually sent to each of the 61 passive participants. Three partially mixed
audio streams are then generated in step 1350. A first partially mixed stream 1
contains audio from speakers 2-3 but not speaker 1. A second partially mixed
stream 2 contains audio from speakers 1-3 but not speaker 2. A third partially
mixed stream 3 contains audio from speakers 1 and 2 but not speaker 3. The first
through third partially mixed audio streams are eventually sent to speakers 1-3
respectively. In this way only four mixed audio streams (one fully mixed and
three partially mixed) need be generated by audio source 1040. This reduces the
work on audio source 1040.
As shown in FIG. 13B, in step 1370, multicaster 1050 replicates packets
in the fully mixed audio stream and set of partially mixed audio streams and
multicasts the replicated packet copies on all of the SVCs (SVC1-SVCN)
assigned to the conference call. NIC 1020 then processes each packet received
on the SVC (step 1380). For clarity, each packet processed internally in
distributed conference bridge 1000 (including packets received at SVCs by NIC
1020) are referred to as internal packets. Internal packets can be any type of
packet format including, but not limited to, IP packets and/or internal egress
packets described above in Figs. 7A and 7B, and the example internal egress or
outbound packet described with respect to FIG. 14A.
For each SVC, NIC 1020 determines whether to discard or forward a
received internal packet for further packet processing and eventual transmission
to a corresponding conference call participant (step 1381). The received internal
packet can be from a fully mixed or partially mixed audio stream. If yes, the
packet is to be forwarded, then control proceeds to step 1390. If no, the packet
is not to be forwarded, then control proceeds to step 1380 to process the next
packet. In step 1390, the packet is processed into a network IP packet. In one
embodiment, packet processor 1070 generates a packet header with at least the
participant's network address information (IP and/or UDP address) obtained from
the look-up table 1025. Packet processor 1070 further adds sequence information
such as RTP/RTCP packet header information (e.g., a timestamp and/or other
type of sequence information). Packet processor 1070 can generate such
sequence information based on the order of received packets and/or based on
sequence information (e.g. the Sequence field) provided in packets generated by
the audio source 1040 (or by multicaster 1050). Packet processor 1070 further
adds a payload in each network packet that includes audio from the received
internal packet being forwarded to a participant. NIC 1020 (or packet processor
1070) then sends the generated IP packet to the participant (step 1395).
One feature of the present invention is that the packet processing
determination in step 1381 can be performed quickly and in real-time during a
conference call. FIG. 13C shows one example routine for carrying out the packet
processing determination step 1381 according to the present invention (steps
1382-1389). This routine is carried out for each outbound packet that arrives on
each SVC. NIC 1020 acts as a filter or selector in determining which packets are
discarded and which are converted to IP packets and sent to a call participant.
When an internal packet arrives on a SVC, NIC 1020 looks up an entry
in look up table 1025 that corresponds to the particular SVC and obtains a CID
value (step 1382). NIC 1020 then determines whether the obtained CID value
matches any CID value in the Total Active Speakers (TAS) field of the internal
packet (step 1383). If yes, control proceeds to step 1384. If no, control proceeds
to step 1386. In step 1384, NIC 1020 determines whether the obtained CID value
matches any CID value in the Included Active Speakers (IAS) field of the internal
packet. If yes, control proceeds to step 1385. If no, control proceeds to step
1387. In step 1385, the packet is discarded. Control then proceeds to step 1389
which returns control to step 1380 to process a next packet. In step 1387, control
jumps to step 1390 for generating an IP packet from the internal packet.
In step 1386, a comparison of the TAS and IAS fields is made. If the
fields are identical (as in the case of a fully mixed audio stream packet), then
control proceeds to step 1387. In step 1387, control jumps to step 1390. If the
TAS and IAS fields are not identical, then control proceeds to step 1385 and the
packet is discarded.
C. Outbound Packet Flow through Distributed Conference Bridge
Outbound packet flow in distributed conference bridge 1000 is described
further with respect to example packets in a 64-person conference call shown in
FIGs. 14 and 15. In FIGs. 14 and 15, mixed audio content in a packet payload is
denoted by a bracket surrounding the respective participants whose audio is
mixed (e.g., {C1,C2,C3}). CID information in packet headers is denoted by
underlining the respective active speaker participants (e.g., C1, C2, C3, etc.).
Sequence information is simply shown by a sequence number 0,1 etc.
In this example, there are 64 participants C1-C64 in a conference call of
which three are identified as active speakers at a given time (C1-C3). Audio
participants C4-C64 are considered passive and their audio is not mixed. Audio
source 1040 generates one fully mixed audio stream FM having audio from all 3
active speakers (C1-C3). FIG. 14B shows two example internal packets 1402,
1404 generated by audio source 1040 during this conference call. Packets 1402,
1404 in stream FM have a packet header and payload. The payloads in packets
1402,1404 each include mixed audio from each of the three active speakers C1-
C3. Packets 1402,1404 each include packet headers having TAS and IAS fields.
The TAS field contains CIDs for the total three active speakers C1-C3. The IAS
field contains CIDs for the active speakers C1-C3 whose content is actually
mixed in the payload of the packet. Packet 1402,1404 further include sequence
information 0 and 1 respectively to indicate packet 1402 precedes packet 1404.
Mixed audio from fully mixed stream FM is eventually sent to each of the 61
currently passive participants (C4-C64).
Three partially mixed audio streams PM1-PM3 are generated by audio
source 1040. FIG. 14B shows two packets 1412, 1414 of first partially mixed
stream PM1. Payloads in packets 1412 andl414 contain mixed audio from
speakers C2 and C3 but not speaker CI. Packets 1412,1414 each include packet
headers. The TAS field contains CIDs for the total three active speakers C1-C3.
The IAS field contains CIDs for the two active speakers C2 and C3 whose
content is actually mixed in the payload of the packet. Packet 1412,1414 have
sequence information 0 and 1 respectively to indicate packet 1412 precedes
packet 1414. FIG. 14B shows two packets 1422,1424 of second partially mixed
stream PM2. Payloads in packets 1422 and 1424 contain mixed audio from
speakers C1 and C3 but not speaker C2. Packets 1422,1424 each include packet
headers. The TAS field contains CIDs for the total three active speakers C1-C3.
The IAS field contains CIDs for the two active speakers C1 and C3 whose
content is actually mixed in the payload of the packet. Packets 1422,1424 have
sequence information 0 and 1 respectively to indicate packet 1422 precedes
packet 1424. FIG. 14B further shows two packets 1432,1434 of third partially
mixed stream PM3. Payloads in packets 1432 and 1434 contain mixed audio from
speakers C1 and C2 but not speaker C3. Packets 1432,1434 each include packet
headers. The TAS field contains CIDs for the total three active speakers C1-C3.
The IAS field contains CIDs for the two active speakers C1 and C2 whose
content is actually mixed in the payload of the packet. Packets 1432,1434 have
sequence information 0 and 1 respectively to indicate packet 1432 precedes
packet 1434.
FIG. 15 is a diagram that illustrates example packet content after the
packets of FIG. 14 have been multicasted and after they have been processed into
IP packets to be sent to appropriate conference call participants according to the
present invention. In particular, packets 1412,1422,1432,1402,1414 are shown
as they are multicast across each of SVC1-SVC64 and arrive at NIC 1020. As
described above with respect to step 1381, NIC 1020 determines for each SVC1-
SVC64 which packets 1412,1422,1432,1402,1414 are appropriate to forward
to a respective conference call participant C1-C64. Network packets (e.g. IP
packets) are then generated by packet processor 1070 and sent to the respective
conference call participant C1-C64.
As shown in FIG. 15, for SVC1, packets 1412 and 1414 are determined
to be forwarded to C1 based on their packet headers. Packets 1412,1414 have
the CID of C1 in the TAS field but not the IAS field. Packets 1412 and 1414 are
converted to network packets 1512 and 1514. Network packets 1512, 1514
include the IP address of C1 (C1ADDR) and die mixed audio from speakers C2
and C3 but not speaker CI. Packets 1512,1514 have sequence information 0 and
1 respectively to indicate packet 1512 precedes packet 1514. For SVC2
(corresponding to conference call participant C2), packet 1422 is determined to
be forwarded to C2. Packet 1422 has the CID of C2 in the TAS field but not the
IAS field. Packet 1422 is converted to network packet 1522. Network packet
1522 includes the IP address of C2 (C2ADDR), sequence information 0, and the
mixed audio from speakers C1 and C3 but not speaker C2. For SVC3
(corresponding to conference call participant C3), packet 1432 is determined to
be forwarded to C3. Packet 1432 has the CID of C3 in the TAS field but not the
IAS field. Packet 1432 is converted to network packet 1532. Network packet
1532 includes the IP address of C3 (C3ADDR), sequence information 0, and the
mixed audio from speakers C1 and C2 but not speaker C3. For SVC4
(corresponding to conference call participant C4), packet 1402 is determined to
be forwarded to C4. Packet 1402 does not have the CID of C4 in the TAS field
and the TAS and IAS fields are identical indicating a fully-mixed stream. Packet
1402 is converted to network packet 1502. Network packet 1502 includes the IP
address of C4 (C4ADDR), sequence information 0, and the mixed audio from all
of the active speakers C1, C2, and C3. Each of the other passive participants C5-
C64 receive similar packets. For example, for SVC64 (corresponding to
conference call participant C64), packet 1402 is determined to be forwarded to
C64. Packet 1402 is converted to network packet 1503. Network packet 1503
includes the IP address of C64 (C64ADDR), sequence infonnation 0, and the
mixed audio from all of the active speakers C1, C2, and C3.
D. Control Logic and Additional Embodiments
Functionality described above with respect to the operation of conference
bridge 1000 (including conference call agent 1010, NIC 1020, switch 1030, audio
source 1040, and multi-caster 1050) can be implemented in control logic. Such
control logic can be implemented in software, firmware, hardware or any
combination thereof.
In one embodiment, distributed conference bridge 1000 is implemented
in a media server such as media server 202. In one embodiment, distributed
conference bridge 1000 is implemented in audio processing platform 230.
Conference call agent 1010 is part of call control and audio feature manager 302.
NIC 306 carries out the network interface functions of NIC 1020 and packet
processors 307 carry out the function of packet processor 1070. Switch 304 is
replaced with switch 1030 and multicaster 1050. Any of audio sources 308 can
carry out the function of audio source 1040.
XI. Conclusion
While specific embodiments of the.present invention have been described
above, it should be understood that they have been presented by way of example
only, and not limitation. It will be understood by those skilled in the art that
various changes in form and details may be made therein without departing from
the spirit and scope of the invention as defined in the appended claims. Thus, the
breadth and scope of the present invention should not be limited by any of the
above-described exemplary embodiments, but should be defined only in
accordance with the following claims and their equivalents.
WE CLAIM :
1. 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 Tn the respective packets.
2. The method as claimed in claim 1, 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 theJnitiated conference call.
3. The method as claimed in claim 2, comprising:
assigning switched virtual circuits (SVCs) to respective participants in the
conference call; wherein said storing step also comprises 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.
4. The method as claimed in claim 1, comprising:
monitoring energy in inbound audio streams of the participants; and
determining a number of active speakers based on the monitored energy.
5. The method as claimed in claim 4, wherein said generating steps (a) and (b)
generate packet headers having active speaker information based on the determined
member of active speakers, and said determining step (d) determines which
malticasted packets to forward to participants based on the active speaker
information in the respective packet headers.
6. The method as claimed in claim 5, 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.
7. The method as claimed in claim 6, 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.
8. The method as claimed in claim 6, wherein said determining step (d)
comprises for each packet being processed at a SVC the steps of:
obtaining a CID value for the SVC; and
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.
9. The method as claimed in claim 8, 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.
10. The method as claimed in claim 6, wherein the packet header comprises
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.
11. The method as claimed in claim 1, 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.
12. The method as claimed in claim 1, 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.
13. The method as claimed in claim 1, comprising mixing audio received over a
network from participants in the conference call who are active speakers.
14. The method as claimed in claim 1, 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.
15. 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 pay load;
a switch; and
a network interface controller,
wherein said switch is coupled between said network interface controller and
said audio source and said switch also 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.
16. The conference bridge as claimed in claim 15, comprising a conference call
agent that initiates a conference call among the participants.
17. The conference bridge as claimed in claim 16, comprising a storage device
that stores conference identifier.information. (CID) and network address information,
associated with each participant in the established conference call.
18. The conference bridge as claimed in claim 17, wherein said storage device
comprises a look-up table.
19. The conference bridge as claimed in claim 17, 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.
20. The conference bridge as claimed in claim 15, 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.
21. The conference bridge as claimed in claim 20, 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.
22. The conference bridge as claimed in claim 21, 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.
23. The conference bridge as claimed in claim 22, wherein said packet headers
generated by said audio source also comprise sequence information.
24. The conference bridge as claimed in claim 15, 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.jn. said set have
payloads which include mixed audio from the at least three active speakers minus
the audio of a respective recipient active speaker.
25. The conference bridge as claimed in claim 15, comprising:
a packet processor that processes packets determined to be forwarded into network
packets having network addresses of participants in the conference call.
26. The conference bridge as claimed in claim 15, wherein said audio source
mixes audio received over a network from participants in the conference call who are
active speakers.
27. The conference bridge as claimed in claim 15, 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.
28. 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.
29. A media server for use in a VOIP 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 also 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.

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.

Documents:

56-kolnp-2004-granted-abstract.pdf

56-kolnp-2004-granted-assignment.pdf

56-kolnp-2004-granted-claims.pdf

56-kolnp-2004-granted-correspondence.pdf

56-kolnp-2004-granted-description (complete).pdf

56-kolnp-2004-granted-drawings.pdf

56-kolnp-2004-granted-examination report.pdf

56-kolnp-2004-granted-form 1.pdf

56-kolnp-2004-granted-form 18.pdf

56-kolnp-2004-granted-form 3.pdf

56-kolnp-2004-granted-form 5.pdf

56-kolnp-2004-granted-gpa.pdf

56-kolnp-2004-granted-reply to examination report.pdf

56-kolnp-2004-granted-specification.pdf

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Patent Number

233928

Indian Patent Application Number

56/KOLNP/2004

PG Journal Number

17/2009

Publication Date

24-Apr-2009

Grant Date

22-Apr-2009

Date of Filing

16-Jan-2004

Name of Patentee

IP UNITY

Applicant Address

475 SYCAMORE DRIVE, MILPITAS, CALIFORNIA

Inventors:

#	Inventor's Name	Inventor's Address
1	DOST SERKAN RECEP	1555 RIVEIRA AVENUE #625, WALNUT CREEK, CA 94596
2	LAURSEN DAVID	4029 SKYLARK LANE, DANVILLE, CA 94506
3	ISRAEL DAVID	2065 CARPENTER STREET, SANTA CLARA, CA 95051
4	MCKNIGHT THOMAS	592 MILL CREEK LANE #304, SANTA CLARA,CA 95054
5	STANWYCK DONALD A.	2182 EAST 129TH DRIVE, THORNTON, CO 80241-1910

PCT International Classification Number

H04L 12/16

PCT International Application Number

PCT/US2002/20359

PCT International Filing date

2002-06-28

PCT Conventions:

#	PCT Application Number	Date of Convention	Priority Country
1	09/893,743	2001-06-29	U.S.A.
2	09/930,500	2001-08-16	U.S.A.
3	10/122,397	2002-04-16	U.S.A.