AU2001100369A4 - Voice-on-demand communication system - Google Patents

Description

AUSTRALIA

Patents Act 1990 COMPLETE SPECIFICATION INOVATION PATENT The following statement is a full description of this invention, including the best method known of preforming it know to me: Real-Time Voice-on-Demand Communication system Field of the invention The present invention relates to computer communication systems and, in particular to Internet Protocol (IP) communication systems, which provide Voice-On-Demand Internet Protocol calling.

The invention is a computer system, which provides IP calling facilities in a software format as opposed to other communication systems, which are all hardware, based.

In recent years the communications industry has observed an increasing demand for versatility in the IP market. The average consumer is less interested in high system performance such as conference calling rather than in everyday usefulness and cost saving ability of an IP system. As a result of this there has been an increasing demand for communication systems and networks which have multimedia and IP capabilities.

Among the most desirable IP capabilities are those associated with zero cost calling facilities and least call routing (LCR) capabilities. A number of uses have been contemplated for transmission of IP traffic. For example a user may want access to music or news on the IP telephones display whilst on the telephone. Also transmission of audio and visual data provides much needed access to valuable information for visually and hearing impaired persons.

IP traffic is routed on communication networks in a very distinct way. Digital signal processing is used to compress data within the source material library so that such data can be transmitted over standard communication links such as a cable, satellite broadcast channel or a standard telephone line receiver specified by subscriber service. The receiver subscriber unit includes a decompressor for decompressing data sent from the source materials library and playing back the decompressed data by means of audio and visual display.

Although known IP communication systems offer significant benefits over copper-cabled communication networks such systems are still subject to high hardware and infrastructure costs.

This Invention relates to the improvements in devices for network communication over Internet Protocol.

The present invention provides a real time, communication-on-demand system, which can be implemented using only the processing capabilities of a CPU, a network hub and an IP capable telephone.

As detailed above, a number of significant difficulties arise when attempting to provide real-time voice-on-demand. It has been found that these difficulties are exacerbated when the Point of Presence (POP) is a server having a set of processors less powerful than 900Mhz of equivalent power built into the routing and switching hardware. Of course higher processors and more sophisticated hardware can be used, but such a network would become prohibitively expensive to outlay and run, and would no be able to offer users price comparable service offered by a traditional communications network. In order to compensate for lack of power, special hardware and other additional capabilities would be needed. The system of the present invention overcomes these difficulties so that real-time voice-on-demand is available to the average consumer on an unmodified communication network.

BRIEF DESCRIPTION OF THE DRAWINGS Diagram 1 shows a simplified high level view of the Voice-on-Demand call system, constructed in accordance with the present invention.

Diagram 2 shows a simple IP phone call where user 1 is calling user 2.

Diagram 3 describes the Voice-on-Demand call messages exchanged for call establishment and how the IP system packages the call messages. The diagram shows the main functional messages of the call process with the Voice-on-Demand system.

Diagram 4 shows an IP phone initiating a call by sending an invite message through the Voice-on-Demand system Diagram 5 is a schematic representation of the method used in accordance with the present invention to manage the flow of data blocks from the Voice-on-Demand system. It depicts the call establishment process where, IP phone 2 responding and setting up a RTP path with IP phone 1.

Diagram 6 shows the general flow of control of the VoD call stack within the server of the present invention.

Diagram 7 is a schematic representation of a call being terminated between the Voice-on- Demand system server and an IP phone.

IN THE DRAWINGS Diagram 1 shows a simplified high level view of the Voice-on-Demand call system, constructed in accordance with the present invention. The system comprises of a network server 100 having a video display 115. The subscribers IP phone 120 connects to the Network Manager module of the server 110 which the routes the call over the Internet or telephone lines 125 via the network hub 130.

In operation, a user calls the desired phone number by means of the IP phone 120. The Voice-on-Demand system network manager 110 transmits a menu of possible selections through to the client and call record module 140. The system then records a log of the call process where the time, date, location, call duration, the IP number of the two phones are stored and logged. The client and call record module the sends the call to the redirect processor 160 where system processors whether the call is to be sent through the Public Switched Telephone Network (PSTN) 180 or the IP network 125.

When the call is sent through the PSTN network 185 the call message is transferred from the redirect processor 160 through to the gateway marshal processor 150 where the call is broken down into a message stack. The message stack (Diagram 6) is then directed through to the PSTN access point (PSTN interface card) 180 and routed through to the dialled phone.

When the call is required to be sent through the IP network or Internet 125 the call message is transferred from the redirect processor 160 through to the implementation and provisioning access point 170 where the call is broken down into a message stack. The message stack (Diagram 6) is then sent through to the network hub 130 by the implementation and provisioning access point 170 where it is transferred to the IP network 125 and routed to its destination.

Diagram 2 shows a simple IP phone connect call over the Voice-on-Demand network. IP phone 1 190 uses the Voice-on-Demand system 100 to connect through to IP phone 2 195.

Diagram 3 describes how the call is initiated by the Voice-on-Demand system 100 in a simplified manner for the call between IP phone 1 190 and IP phone 2 195. To begin the call process IP phone 1 190 dials the corresponding number of IP phone 2 195 to initiate the call.

IP phone 1 190 connects to the Voice-on-Demand systems Network manager module 110 where the IP phone number initiating the call is logged. The Call is then transferred to the Client and call record Processor module 140 where the call time log begins, and the time, date, location and IP numbers of the phones are logged and stored. Once details are logged the client and call record processor 140 sends an initiate command to the Redirect processor module 160. Because the call is being routed through an IP network the redirect processor 160 initiates the call where it is then processed at the implementation and provisioning access point module 170. The call is then processed into a message stack (Diagram 6) and the system initiates the ringing command to IP phone 2 195. A 100, Ringing message is then returned to the Network Manager module 110 to show the connection over the network has been made. The call is then answered by IP phone 2 195 and a 200, OK message is sent to the Client and call record processor module 140 where the call timer begins. Once the connect has been initiated the Network Hub 130 begins transferring the call message stacks (diagram 6) over the IP network. Call message stacks (diagram 6) are then sent over the IP network where a two way voice channel is established over real-time transfer protocol (RTP) and the conversation takes place between user 1 and 2 over the IP network. User 2 then hangs up which sends an END command to the Client and call record processor module 140 of the Voice-on-Demand system which ends the call log and saves the details in the systems memory banks. User 1 on IP phone 1 190 hangs up which directs the Client and call record processor module to send a 200, OK message to the Network manager to acknowledge that the call has been terminated. (See table 3 for simplified call flow steps) Table 4 shows the call request and response messages sent by the Voice-on-Demand system during a call over an IP network.

As will be described in greater detail below, diagram 4 through to diagram 9 describe and show a high level view of how a call is initiated, established and terminated on the Voice-on- Demand Network in a closed network run through an inter-office network environment.

A more detailed diagram of the call initiation in the Voice-on-Demand system 100 of the present invention is depicted in Diagram 4. IP phone 1 200 and IP phone 2 210 are on the local IP Network (inter-office network). IP phone 1 200 sends an INITIATE message intended for IP phone 2 210. The INITIATE message is received by the Marshal Server Module 1 (MSM1) 220. The Voice-on-Demand system authenticates the user and forwards the INTIATE message to the Redirect Server (RS) 240. This is the normal routine (The Marshal Processor Module 1 220 forwards all INITIATE messages from all authorised users to the Redirect Server 240) The Redirect Server 240 responds by sending a 200, Voice message to the Marshal Processor Module 1 220. The 200, Voice message informs the Marshal Processor Module 1 220 that the INITIATE message was intended for the Redirect Server 240. The 200, Voice message also provides the routing information that enables the Marshal Processing Module 1 220 to forward the INITIATE towards its intended destination. The Marshal Processor Module 1 220 sends an Real-time Transfer Protocol (RTP) message back to the Redirect Server 240 acknowledging receipt of the 200, Voice message. The Marshal Processor Module 1 220 then forwards the Initiate message to the Marshal Processor Module 2 230 which is the proxy server for the intended destination. Marshal Processor Module 2 230 forwards the INITIATE message to the Redirect Server 240, which responds with a 200, Voice message and sends it to Marshal Processor Module 2 230. Marshal Processor Module 2 230 then responds to Marshal Processor Module 1 220 RTP message by sending one of its own (an RTP message) back to the Redirect Server 240 acknowledging receipt of the 200, Voice message, and then forwards the INITIATE message to IP phone 2 210. This is described systematically in diagram 4, with table 5 showing the interaction and description of the steps taken for call initiation in diagram 4.

Call establishment is shown in a high level view in diagram 5 where IP phone 2 210 responds to call initiation by setting up an RTP (Real-time Transfer Protocol) path 250 with IP phone 1 200. To begin the call establishment process IP phone 2 210 sends a 100, Ringing, response to Marshal Processor Module 2 230 which is forwarded to Marshal Processor Module 1 220 and then forwarded to IP phone 1 200. IP phone 1 200 picks up which sees IP phone 2 210 send a 200, OK message to the Marshal Processor Module 1 220. This means that the phone has been activated and is ready to establish voice channel contact with IP Phone 1 200. IP phone 1 200 sends a Voice message confirming that it is ready to connect a voice channel. The pick up has now been acknowledged. A voice channel is established using Real-time Transfer protocol (RTP) 250 enabling a conversation to take place. This is described systematically in diagram 5, with table 6 showing the interaction and description of the steps taken for call initiation in diagram 4.

The voice channel established by the Real-time Transfer Protocol (RTP) uses message stacks to transfer voice packets over the network. Diagram 6 shows how the components of the voice stack work together to digitally compress and decompress the data ready for transfer over the network. Message stacks are formed in the Marshal Processor Module (MPM) where voice comes from either IP phone. When voice traffic arrives at the MPM from the transmitting IP phone it is forwarded to the Digital Compressor 310 to be digitally compressed. The digitally compressed call stack is then forwarded to the sending transceiver 320 where the call stack is broken down into to small data packets ready for transmission. A record of the call stack is logged in the Transaction Database 340 before it is forwarded to the RTP access point 360 for transmission to the receiving IP phone.

When an IP phone is receiving voice the incoming data is received by the MPM and forwarded to the RTP access point 360 where a log of its receipt is made by the Transaction Database 340. The receiving transceiver 330 is then forwarded the data packets for construction into the call stacks. When the call stack is reconstructed the Retransmitter Filter 350 filters it. The filtered call stack is then forwarded to the Digital Compressor 310 for decompression before being transferred to the receiving IP phone.

Diagram 7 demonstrated the call termination procedure of the current invention. When the conversation is over, both phones hang up. The first phone to hang up sends an END message to the other IP phone. This END message terminates the Real-time Transfer Protocol (RTP) for that phone. The other IP phone responds with a 200, OK message and terminates its side of the RTP path.

When IP phone 2 210 terminates (hangs up first) it sends an END request, through to the Marshal Processing module 2 230 which is directed through to Marshal Processing module 1 220 and through to IP phone 1 200. IP phone 1 200 sends a 200, OK message, through the Marshal Processor Modules 1 220 and 2 230 to IP phone 2. The END and OK message s trigger the voice channel to shut down.

Claims (5)

1. The Voice-on-Demand system is a distributed network of module processors that provide Voice over Internet Protocol (VolP) telephony services. Voice-on-Demand supports devices that communicate Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP) or H.323 messages. Voice-on-Demand also supports analog telephones via residential gateways.

2.Voice-on-Demand supports on-network and off-network calling. Off-network calling enables subscribers to connect to parties through either the Internet or the Public Switched Telephone Network (PSTN). See figure 1

3.The Voice-on-Demand system appears as an assembly of basic components. These components are described below in Table 1 Table 1. System Components Component Description Voice-on-Demand This is the telephony application. Figure 1 shows System an abstract representation of the Voice-on- Demand system server modules Protocols The Voice-on-Demand system uses several protocols to communicate between its components. The call signaling processes use IP messaging to communicate internally within the Voice-on-Demand system and externally with gateways and IP phones. The graphical user interface (GUI) enables technicians to provision the system, and administrators to set up users and monitor the GUI system's performance. The GUI is web-enabled and requires a Java plug-in to run in a web browser IP Phone Voice-on-Demand supports a variety of phone appliances including IP phones and IP IP Phone User Agent (UA) software applications. IP phones may be connected to the Voice-on- Demand system over any IP network. Processors Table 1-1. Voice-on-Demand System Components (Continued) Component Description Gateways The gateways not only provide entry points between networks; they also provide translation between IP-based networks and other network types. The Voice-On-Demand system works with two types of gateways, the residential gateway and the PSTN gateway. Residential Gateway Residential gateways translate analog signals into IP packets, to permit subscribers with analog phone sets/devices to make and receive IP-based calls. PSTN Gateway PSTN gateways permit IP-based networks to exchange calls with end-points on the PSTN, by providing translation between SIP messages and one of these signal types: Analog Channel Associated Signaling (CAS) Primary Rate Interface (PRI) Table 1-2 describes the Processor modules included in the VoD system. Table 1-2. Voice-on-Demand Processor Modules Processor Description Modules Marshal The Marshal Processor (MP) is an Processor implementation of the IP proxy server and Module act as the initial point of contact for all IP signals that enter the VoD system. The MP provides authentication, forwarding, call stack breakdown, digital compression and system response commands. Redirect The Redirect Processor (RP) is a combined Processor implementation of the IP redirect, Module registration and location modules. The RP stores contact and feature data for all registered subscribers and a dialing plan to enable routing for off network calls. Client and Call The Call Record (CR) processor receives Record Module call data from the Marshal Processor and formats it into data that can be transmitted to third party billing systems for invoicing. Network The Network Manager provides the Manager administrator with the ability to monitor the system through Simple Network Management Protocol (SNMP) messages. Table 1-2. Voice-on-Demand Processor Modules (Continued) r. I- Processor Modules Implementation and Provisioning Processor Description Description The Implementation and provisioning Processor stores data records about each system user and processor module, and distributes this information throughout the system via a subscribe-notify model. It also monitors the flow of pulsing signals emitted by the other processors, and provides information about to the flow of heartbeats to the GUI. This information helps the System Administrator know if the server modules are up or down. The module has been designed to use Open Policy Service (OPS) to provide Quality of Service (QoS) bandwidth reservation for calls or call segments that are transmitted over the Internet. The module is Also capable of using Open Settlement Protocol (OSP) to interact with clearinghouses for reserving bandwidth and authorising the use of a network for inter- network calls. The Voice-on-Demand system of claims 1 to 3 wherein the entire communication system is located in two hardware units, a central processing unit (Network Server) and a network hub.

6. The Voice-on-Demand system device of claim 1 wherein the system modules are software components, which decipher communication, protocols.

7. A Voice-on-Demand system substantially as herein described with reference to the accompanying diagrams. Tell Corporation 12 August 2001