Patent Application
20060251052
The present invention relates generally to communication networks and, more particularly, to a method and apparatus for enabling local survivability during network disruptions using premise based solutions, e.g., VoIP solutions, in packet networks, e.g. Voice over Internet Protocol (VoIP) networks.
Large enterprises have traditionally relied on network carriers to provide many of their telephony services. As telephony services become more IP centric and network intelligence is distributed more to endpoint components, enterprises are re-examining their relationships with carriers. Some enterprise customers are choosing to rely primarily on premise based VoIP solutions to support intra-company voice communications and use carrier network services only for Internet Protocol (IP) transport between company locations. These premise based VoIP solutions preclude access to innovative VoIP network services and are costly and difficult to operate and maintain.
Therefore, a need exists for a method and apparatus for enabling local survivability during network disruptions using premise based solutions, e.g., VoIP solutions in packet, networks, e.g., VoIP networks.
In one embodiment, the present invention enables a network-based solution for enterprise customers, where the solution is integrated with premise-based equipment, e.g., VoIP equipment. The premise-based solution can be quickly enabled in the event that the primary VoIP network connectivity is lost. The premise-based equipment, e.g. a VoIP switch, can be minimally configured and equipped to play the role of a hot standby proxy for network based VoIP services if connection to the primary VoIP network is lost. Intelligent services typically accessed in network based VoIP services can be enabled on the premise equipment and VoIP gateways to the PSTN can be activated. Network operators can remotely activate this equipment on behalf of the enterprise customer until the primary VoIP network connection is restored.
The teaching of the present invention can be readily understood by considering the following detailed description in conjunction with the accompanying drawings, in which:
To facilitate understanding, identical reference numerals have been used where possible, to designate identical elements that are common to the figures.
To better understand the present invention,
In one embodiment, the VoIP network may comprise various types of customer endpoint devices connected via various types of access networks to a carrier (a service provider) VoIP core infrastructure over an Internet Protocol/Multi-Protocol Label Switching (IP/MPLS) based core backbone network. Broadly defined, a VoIP network is a network that is capable of carrying voice signals as packetized data over an IP network. The present invention is described below in the context of an illustrative VoIP network. Thus, the present invention should not be interpreted to be limited by this particular illustrative architecture.
The customer endpoint devices can be either Time Division Multiplexing (TDM) based or IP based. TDM based customer endpoint devices 122, 123, 134, and 135 typically comprise of TDM phones or Private Branch Exchange (PBX). IP based customer endpoint devices 144 and 145 typically comprise IP phones or PBX. The Terminal Adaptors (TA) 132 and 133 are used to provide necessary interworking functions between TDM customer endpoint devices, such as analog phones, and packet based access network technologies, such as Digital Subscriber Loop (DSL) or Cable broadband access networks. TDM based customer endpoint devices access VoIP services by using either a Public Switched Telephone Network (PSTN) 120, 121 or a broadband access network via a TA 132 or 133. IP based customer endpoint devices access VoIP services by using a Local Area Network (LAN) 140 and 141 with a VoIP gateway or router 142 and 143, respectively.
The access networks can be either TDM or packet based. A TDM PSTN 120 or 121 is used to support TDM customer endpoint devices connected via traditional phone lines. A packet based access network, such as Frame Relay, ATM, Ethernet or IP, is used to support IP based customer endpoint devices via a customer LAN, e.g., 140 with a VoIP gateway and router 142. A packet based access network 130 or 131, such as DSL or Cable, when used together with a TA 132 or 133, is used to support TDM based customer endpoint devices.
The core VoIP infrastructure comprises of several key VoIP components, such the Border Element (BE) 112 and 113, the Call Control Element (CCE) 111, and VoIP related servers 114. The BE resides at the edge of the VoIP core infrastructure and interfaces with customers endpoints over various types of access networks. A BE is typically implemented as a Media Gateway and performs signaling, media control, security, and call admission control and related functions. The CCE resides within the VoIP infrastructure and is connected to the BEs using the Session Initiation Protocol (SIP) over the underlying IP/MPLS based core backbone network 110. The CCE is typically implemented as a Media Gateway Controller and performs network wide call control related functions as well as interacts with the appropriate VoIP service related servers when necessary. The CCE functions as a SIP back-to-back user agent and is a signaling endpoint for all call legs between all BEs and the CCE. The CCE may need to interact with various VoIP related servers in order to complete a call that require certain service specific features, e.g. translation of an E.164 voice network address into an IP address.
For calls that originate or terminate in a different carrier, they can be handled through the PSTN 120 and 121 or the Partner IP Carrier 160 interconnections. For originating or terminating TDM calls, they can be handled via existing PSTN interconnections to the other carrier. For originating or terminating VoIP calls, they can be handled via the Partner IP carrier interface 160 to the other carrier.
In order to illustrate how the different components operate to support a VoIP call, the following call scenario is used to illustrate how a VoIP call is setup between two customer endpoints. A customer using IP device 144 at location A places a call to another customer at location Z using TDM device 135. During the call setup, a setup signaling message is sent from IP device 144, through the LAN 140, the VoIP Gateway/Router 142, and the associated packet based access network, to BE 112. BE 112 will then send a setup signaling message, such as a SIP-INVITE message if SIP is used, to CCE 111. CCE 111 looks at the called party information and queries the necessary VoIP service related server 114 to obtain the information to complete this call. If BE 113 needs to be involved in completing the call; CCE 111 sends another call setup message, such as a SIP-INVITE message if SIP is used, to BE 113. Upon receiving the call setup message, BE 113 forwards the call setup message, via broadband network 131, to TA 133. TA 133 then identifies the appropriate TDM device 135 and rings that device. Once the call is accepted at location Z by the called party, a call acknowledgement signaling message, such as a SIP-ACK message if SIP is used, is sent in the reverse direction back to the CCE 111. After the CCE 111 receives the call acknowledgement message, it will then send a call acknowledgement signaling message, such as a SIP-ACK message if SIP is used, toward the calling party. In addition, the CCE 111 also provides the necessary information of the call to both BE 112 and BE 113 so that the call data exchange can proceed directly between BE 112 and BE 113. The call signaling path 150 and the call data path 151 are illustratively shown in
Note that a customer in location A using any endpoint device type with its associated access network type can communicate with another customer in location Z using any endpoint device type with its associated network type as well. For instance, a customer at location A using IP customer endpoint device 144 with packet based access network 140 can call another customer at location Z using TDM endpoint device 123 with PSTN access network 121. The BEs 112 and 113 are responsible for the necessary signaling protocol translation, e.g., SS7 to and from SIP, and media format conversion, such as TDM voice format to and from IP based packet voice format.
Large enterprises have traditionally relied on network carriers to provide many of their telephony services. As telephony services become more IP centric and network intelligence is distributed more to endpoint components, enterprises are re-examining their relationships with carriers. Some enterprise customers are choosing to rely primarily on premise based solutions, e.g., VoIP solutions, to support intra-company voice communications and use carrier network services only for Internet Protocol (IP) transport between company locations. These premise based VoIP solutions preclude access to innovative VoIP network services and are costly and difficult to operate and maintain. However, these premise based solutions, when used in conjunction with network based VoIP services, can enable local survivability, e.g., intra-company voice communication survivability, in the event of VoIP network failures.
To address this need, in one embodiment the present invention enables a network-based solution for enterprise customers that is integrated with premise-based VoIP equipment. The premise-based solution can be quickly enabled in the event that the primary VoIP network connectivity is lost. The premise-based equipment, e.g. a VoIP switch, can be minimally configured and equipped to play the role of a hot standby proxy for network based VoIP services if connection to the primary VoIP network is lost. Intelligent services typically accessed in network based VoIP services can be enabled on the premise equipment and VoIP gateways to the PSTN can be activated. Network operators can remotely activate this equipment on behalf of the enterprise customer until the primary VoIP network connection is restored.
In one example, a VoIP service disrupting event 260 has prevented the use of the primary VoIP network 210 to complete calls between the two company locations. Once the service disruption has been detected, the network provider of the VoIP network 210 immediately activates premise based VoIP switches, 251 and 252, to ensure continuous voice communication between the two locations. For example, intra-enterprise calls are redirected by premise based Internet Protocol routers 242 and 243 located at different locations of the enterprise to its corresponding premise based switch. Voice communications between the two locations can be supported using communication path 261 via the corporate WAN 220 and/or using communication path 262 via PSTN 230. Once the VoIP service disrupting event 260 is cleared, the voice communications between the two locations can again then be carried over VoIP network 210 by deactivating the two premise based VoIP switches.
In step 310, the method 300 detects an outage of the primary packet network, e.g., a VoIP network. In step 320, the method activates all premise based switches, e.g., VoIP switches, to enable backup voice communications using the premise based solution, e.g., premise based VoIP solution. In step 330, the method routes all calls to the premise based VoIP switches. In step 340, the method processes all intra-company calls using the premise based VoIP solution over corporate WAN and/or PSTN until the disruption is cleared. The method ends in step 350.
It should be noted that the present invention can be implemented in software and/or in a combination of software and hardware, e.g., using application specific integrated circuits (ASIC), a general purpose computer or any other hardware equivalents. In one embodiment, the present local survivability premise based solution module or process 405 can be loaded into memory 404 and executed by processor 402 to implement the functions as discussed above. As such, the present local survivability premise based solution process 405 (including associated data structures) of the present invention can be stored on a computer readable medium or carrier, e.g., RAM memory, magnetic or optical drive or diskette and the like.
While various embodiments have been described above, it should be understood that they have been presented by way of example only, and not limitation. Thus, the breadth and scope of a preferred embodiment 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.
