Abstract: The Federal Communications Commission Order 05-116, E911 Requirements for IP-Enabled Service Providers, (FCC Order) requires interconnected Voice over Internet Protocol (VoIP) service providers to deliver all emergency service calls to a VoIP service user's local emergency service operator as a standard feature of the service. Under the FCC Order, VoIP service providers must supply emergency service if the VoIP service user accesses the VoIP service over broadband and if the VoIP service provider interconnects to and from the Public Switched Telephone Network (PSTN). Embodiments of the present invention guarantee compliance with the FCC Order by routing a call as either a VoIP call over the Internet or as a conventional call over the PSTN, verifying that at least one of the VoIP or PSTN telephony connections supports emergency service and routing calls over that connection or, if there is no connection supplying emergency service, restricting calling over the VoIP connection.

Abstract: A method and apparatus for bundling packets together for transmission in a Voice over IP communications network based on packet location within a talk spurt. Illustratively, all frames other than the first and last frames of a talk spurt may be advantageously bundled up to a predetermined maximum bundle size. This results from the recognition that the first and last packets of the talk spurt are the packets that will most directly affect the conversational delay. Therefore, other packets can be advantageously considered to be “non-critical” (with respect to conversational delay), and thus, may be bundled together with one or more other packets. In this manner, bandwidth may be advantageously reduced without negatively impacting the perceived conversational delay.

Abstract: An exemplary method provides ringback information to a calling party device. A call establishment request is received in an Internet Protocol (IP) Multimedia Subsystem (IMS) network from the calling party device by a session initiation protocol, SIP, application server, SIP-AS. Ringback information associated with the called party, which may require condition evaluation such as time-of-day, day-of-week or may be unconditionally determined, is identified in response to the call establishment request. A push transmission technique is used by the SIP-AS to transmit the ringback information via one or more bearer packets sent from the SIP-AS to the calling party device.

Abstract: The present invention provides a system, method and apparatus for providing network level and nodal level vulnerability protection in VoIP networks by receiving a communication, filtering the received communication using three or more stages selected from the group comprising a media protection and filtering plane, a policy based filtering plane, a signature based filtering plane, a protocol anomaly detection and filtering plane and a behavioral learning based filtering plane, and either allowing or denying the received communication based the filtering step. The stages are applicable to one or more protocols including SIP, IMS, UMA, H.248, H.323, RTP, CSTA/XML or a combination thereof. In addition, the stages can be implemented within a single device or are distributed across a network (e.g., SIP network, a UMA network, an IMS network or a combination thereof).

Abstract: Provided is an apparatus and method for processing a Voice over Internet Protocol (VoIP) packet with multiple frames. The apparatus includes: a transmission packet processing unit for receiving a frame from a speech codec, forming a Real-time Transport Protocol (RTP) payload in a form of multiple frames and transmitting the RTP payload to an RTP stack; and a reception packet processing unit for receiving an RTP packet from the RTP stack, storing the RTP packet in a jitter buffer, performing dejittering, separating frames from the RTP payload one by one and transmitting the frames to the speech codec.

Abstract: A method for determining the location of a VoIP caller includes receiving an emergency call from a VoIP phone from the internet via a VoIP enabling device. A device identifier of the VoIP enabling device through which the call from the VoIP device originated is received. The device identifier is compared against a pre-defined table of device identifiers, where each device identifier in the table has an associated physical location. The physical location of the VoIP enabling device is returned during the call from the VoIP phone.

Abstract: A method, system and apparatus for announcing caller information to users over a television link. The method comprises receiving, via a portion of a data network adapted for handling Voice-over-IP (VoIP) calls, Session Initiation Protocol (SIP) signalling information associated with an incoming call destined for a destination party, the SIP signalling information comprising a portion conveying information regarding an origin of the incoming call and information regarding an identity of the destination party. The method further comprises attempting to establish a telephony link with a device associated with the destination party over a communication network in response to the receiving. The method further comprises releasing the portion of the SIP signalling information to a processing entity to cause delivery of an indication of the origin of the incoming call over a television link associated with the destination party, the indication based on the information regarding an origin of the incoming call.

Abstract: Devices, softwares and methods send a warning signal when a periodic signal is present in the audio content of a VoIP connection. In response to the warning signal, an acoustic echo canceller of an IP telephone discontinues its adapting process. A signaling network switch detects when a periodic signal is present in the audio content, and issues a warning signal. A signaling call manager device, when it transmits a tone that has periodicity, it also transmits a warning signal.

Abstract: A software suite is disclosed for routing communication events over a data-packet-network using an IP session initiation and management protocol. The software suite comprises, a server application running on the network for computing and serving routing determinations per request, a session management application running on the network for initiating and managing routed and established session events, a parsing application running on the network for parsing request data received under session initiation protocol and a conversion application running on the network for converting data received under session initiation protocol into a routing request. All received communication requests for routing are in the form of the session initiation protocol wherein they are parsed and converted into routing requests processed by the server application and routed to determined destinations and wherein events are established as session events conducted under the session initiation and management protocol.

Abstract: A method of managing Voice over Internet Protocol (VoIP) calls with an Internet Protocol Private Branch Exchange (IP PBX) is provided, including: receiving at an IP PBX device an incoming call directed to a target telephone number, said IP PBX device storing a contact list parameter including one or more rule fields, each rule field including a telephone number identifier and an associated target extension identifier; searching the contact list parameter to identify the telephone number identifier corresponding to the target telephone number; and routing the incoming call to one or more target extensions corresponding to the target extension identifier associated with the identified telephone number identifier.

Abstract: VoIP Routing Registry (VRR) has been created to allow VoIP Service Providers to direct calls properly over their IP networks. In operation, the VRR translates a 10-digit or 6-digit telephone number into a URI that can be used for routing. The VRR must also take into consideration Local Number Portability (LNP).

Abstract: Disclosed above are various embodiments of VoIP communication systems that utilize low cost IP phones that rely on a centralized VoIP controller for much of the processing. Reducing the processing taking place on an IP phone may reduce the number of components that need to be on the IP phone which may result in a less expensive IP phone in terms of both cost and power. When the IP phone is embodied as a WIPP, the reduced processing may also result in more efficient communication between the WIPP and an AP. The increased communication efficiency may result in less power being used by the WIPP and effectively extend the battery life. Since a number of redundant components have been centralized, the VoIP system as a whole may be less costly. Also, centralized control may provide greater functionality and versatility in the setup and configuration of a VoIP communication system.

Abstract: A telephone relay system enabling a terminal having multiple telephone numbers to originate and receive calls same as a terminal having only one telephone number is provided. When a connection management unit of an agent device receives a call request from a terminal device, it activates a terminal calling unit while designating a current telephone number and also activates a counterpart calling unit to establish a terminal-side connection between the current telephone number of the terminal device and the agent device and a counterpart-side connection between the agent device and the counterpart terminal. When an incoming call from the counterpart terminal arrives, the connection management unit establishes a counterpart-side connection corresponding to the incoming call, and activates the terminal calling unit while designating the current telephone number to establish a terminal-side connection.

Abstract: A voice over Internet hub that uses existing audio or video systems to handle phone calls. The hub is linked to the audio system and produces an audio output to the audio system. The audio system may be a stereo system or television. The hub generates audio signals for ring tones, key clicks and voice signals. The hub may also be linked to a video system. The hub is controlled by a remote control unit having a microphone and loudspeaker.

Abstract: One embodiment of a call handling method, among others, includes receiving a query from a switching system indicating that a called party telephone is busy; examining handling information for the called party to determine if the called party has voice over Internet Protocol (VoIP) service; and instructing the switching system to then route the analog call over a VoIP connection to the called party after the called party chooses to communicate using the VoIP service.

Abstract: The present invention discloses a method and a system for implementing communication for a CSI terminal, the system includes: an IP Multimedia Subsystem Terminal Module (IMS TM), a Circuit Switched Terminal Module (CS TM), a CS/IMS interworking gateway. The method includes: the IMS IM indicates the CS TM to establish a CS/IMS interworking call connection from the CS TM to the IMS TM, and establish a CS bearer between the CS TM and CS/IMS interworking gateway, the IMS TM correlated controls the two sessions respectively with the CS/IMS interworking gateway and the peer side, so as to make the media components it exchanged with the peer side be exchanged between the peer side and the CS/IMS interworking gateway via IP bearer, and then be exchanged between the CS/IMS interworking gateway and the CSI terminal through the established CS bearer. Thus the present invention ensures the communication of the CSI terminal.

Abstract: According to one embodiment, a telephone system includes a memory which stores a group table in which a plurality of groups divided and configured by using terminal IDs corresponding to the plurality of interface units are associated with interface units belonging to the groups, and a controller which refers to the group table when connection requests arrive from the IP networks or IP telephone terminals via the control channels and also when the communication channels are selected, preferentially selects idle communication channels in the group to which the control channels with the connection requests arrived thereat based on the reference result, and connects the IP networks or the IP telephone terminals that are request origins via the selected communication channels.

Abstract: A method of operating a virtual private network includes the steps of: (a) dialing a custom number; (b) receiving the custom number at a switch (20) of a first carrier; (c) routing the call to a switch (18) of a second carrier based on having received the custom number; and (d) when the call to the switch (18) of the second carrier includes a plain old telephone number translation of the customer number, routing the call over the second carrier to a service switching point (26) of a third carrier.

Abstract: A communications system multiplexes voice communications signals onto one or more transport level connections established through a packetized network, such as the Internet. The invention supports the use of variable-length packets and accommodates variable jitter. The system conforms to real time protocol (RTP) and employs internet telephone gateways (ITGs) to bind users to channel identifiers, to indicate payload type and length, to provide channel identification and time stamps, and to indicate cessation and resumption of voice traffic from a particular user through use of, for example, marker bits.

Abstract: Techniques for distributing digital content include receiving provider content over a network connection at a customer premises node located on premises of a first customer. The provider content is offered by a network service provider different from the first customer. Provider data based on the provider content is stored in non-volatile storage on the customer premises node of the first customer. It is determined whether conditions are satisfied for sending the provider content to a second customer different from the first customer. If it is determined that such conditions are satisfied, then the provider data is retrieved from the non-volatile storage, and data based on the provider data is sent over the network connection for receipt by the second customer. Thereby a customer premises node serves as a cache of provider content for other customer premises nodes on the same last mile segment or access network.

Abstract: A communication equipment for an Internet Protocol communication network including a set of interfaces each connected to one or more other communication equipments and having means for receiving an address prefix from a first other communication equipment. It further includes allocation means for allocating to at least a portion of the other communication equipments to which the first other equipment does not belong a sub-prefix formed of the address prefix concatenated with an individual identifier whose length depends on the total number of other communication equipments.

Abstract: A communication terminal and a method for initiating a communication link from a communication terminal to a remote party, where the method includes sensing input of an address number for the remote party. Upon sensing the input of an address number a plurality of selectable items, representing different communication channels usable for setting up a communication link from the terminal, are presented on a display of the terminal. A navigation tool can be used to change selected communication channel, such as from voice call to messaging, after which setting up of a communication link using the selected communication channel can proceed.

Abstract: A technique is provided for implementing deep-packet inspection (DPI) services in a MPLS/VPN configured computer network. The technique employs a novel self-learning algorithm that analyzes data packets belonging to different unidirectional tunnels in the MPLS/VPN network and determines whether the analyzed data packets transport data in the same VPN. If so, the unidirectional tunnels containing the analyzed data packets are associated with a common layer-2 identification (L2ID) value. Unlike conventional flow-classification procedures, the inventive technique classifies a data packet by first associating the data packet with a L2ID value and then classifying the packet as belonging to a particular data flow based on a novel 6-tuple consisting of a conventional 5-tuple plus the packet's L2ID value.

Abstract: A system is provided that includes a hardware forwarding engine that routes data packets. The system also comprises a processor coupled to the hardware forwarding engine, the processor having a software forwarding engine that routes data packets. If data packets are forwarded from the hardware forwarding engine to the software forwarding engine, the hardware forwarding engine provides route lookup results for at least some of the data packets forwarded to the software forwarding engine.

Abstract: Disclosed are mechanisms for facilitating the use of DHCP (dynamic host configuration protocol) binding data. In general, certain applications include mechanisms for intercepting data being sent from a node and then determining whether the data corresponds to a valid IP address and MAC address binding. Embodiments of the present invention provide mechanisms for sharing such DHCP binding data between routers (or other type of network devices) in a redundancy group so that any of the routers may take over the data inspection to validate DHCP bindings. In particular aspects of the invention, the DHCP binding data is validated in procedures related to DHCP snooping, dynamic ARP (address resolution protocol) inspection, and the like.

Abstract: A method and system for routing a data packet, intended for a destination node, from a source node to a first intermediate node in a network is useful for improving network efficiency and participation. The method includes determining a priority level of the source node and a priority level of the first intermediate node (step 505). A routing path is then selected that routes the data packet from the source node to the first intermediate node based on a comparison of the priority level of the source node and the priority level of the first intermediate node (step 515). The priority level of the source node is determined in part by a number of previous data packets that were routed through the source node, and the priority level of the first intermediate node is determined in part by a number of previous data packets that were routed through the first intermediate node.

Abstract: A communications system, such as a Digital Subscriber Line Access Multiplexer (DSLAM), or corresponding method increases available bandwidth for transmission of data, video, and audio to a customer premise, or curb node for further distribution to customer premises, within a network. In one embodiment, a system comprises a host digital terminal (HDT) including an Ethernet switch unit and multiple optical interface units coupled via a bus. The optical interface units are configured to communicate over an optical communications link with broadband cards of optical network units (ONUs). The ONUs also include data cards coupled to the broadband cards via a bus. The data cards are configured to communicate over end user communications links to end user nodes.

Abstract: Aspects of a method and system for processing audio and video transport may comprise synchronizing streams with associated program content based on 24-bit presentation time stamps (PTSs) and 32-bit system clock samples (SCSs) field without using decoding time stamps (DTSs). A PTS field may be provided as part of a packetized elementary stream (PES) packet syntax. An SCS field may be provided as part of a transport stream (TS) syntax. A system clock (STC) may be determined from received SCS values. Video and/or audio data may be decoded as soon as they arrive and when there is at least one display and/or presentation frame buffer available. The video and/or audio data may be presented for display and/or play based on a comparison of the PTS and STC values. When a frame does not have an associated PTS, a PTS extrapolation operation may generate an extrapolated PTS value from previously received PTSs.

Abstract: Integrated active-active fibre channel capability in SAS and SATA is disclosed. In one embodiment, a system includes a data processing system to communicate a fibre channel frame through a network and a conversion module in a storage device coupled to the network to translate between the fibre channel frame and one or more of a SATA frame and a SAS frame. The conversion module may further include an active-active module to provide multiple paths from the data processing system to the storage device, a context associated with one or more outstanding commands, and a mapping module to translate between a logical block address and a logical block address count for one or more of a 520 block, a 524 block, and a 528 block SCSI command and a corresponding logical block address and a corresponding logical block address count for a 512 block SATA command.

Abstract: A method for processing SIP requests in an IMS network includes: an AS in the IMS network receiving a first SIP request forwarded by a S-CSCF entity, generating a second SIP request and deciding whether it is needed to associate the second SIP request with the first SIP request in terms of service logic at the S-CSCF entity, if it is needed to associate two requests, removing the URI of the AS from the Route header of the first SIP request; otherwise, regenerating the Route header of the second SIP request in the originating UA behavior mode and sending the second SIP request to the S-CSCF entity. Also disclosed is a system for processing SIP requests in an IMS network, including an AS and a S-CSCF entity. Also disclosed is an AS for processing SIP requests in an IMS network and a network entity for service control.

Abstract: A method and system detects a network connection in a communications system, such as a narrowband communications system, using Virtual Local Area Network (VLAN) identification. In one embodiment, a first node transmits a message to a specific second node among a group of second nodes. The message from the first node includes a source Medium Access Control (MAC) address, a broadcast address, and a unique VLAN identification corresponding to a port on the first node. The specific second node processes the message, and then transmits its own MAC address to the first node, along with the unique VLAN identification received in the original message from the first node. The first node then updates stored information about the second node and uses the information in future communications to the second node.

Abstract: A method and system generates a network quality clock signal in a communications system by synthesizing a first clock signal based on arrival rate of packets transmitted via a network link at a rate according to a network clock. The system then synthesizes a second clock signal is then synthesized based on the first clock signal. In one embodiment, the second clock signal has a frequency substantially the same as the network clock.

Abstract: A method and device for creating/reproducing a multimedia file, and an information recording medium recording the multimedia file are provided. According to the method for creating the multimedia file, multimedia raw data is coded in accordance with characteristics of a device reproducing the multimedia raw data. After that, header information containing device identification (ID) information of the device is attached to the coded multimedia data, so that the multimedia file is created.

Abstract: A method and a computer program product embodying software for measuring the utilization of all components in an end-to-end application and automatically associating them with the end-user and end-user activity. One aspect of the invention deals with the measurement and automatic association of server application utilization by the client and the association of the utilization with the end-user and end-user activity. Another aspect of the invention deals with the automatic association of downstream transactions with the client request. Yet another aspect of the invention uses the first two aspects to provide automatic association of utilization measurements of all components in an end-to-end application.

Abstract: A method and integrated hardware system for multicast transmission shaping in EPONs which employs a selective multicast algorithm that significantly reduces the complexity of traffic shaping for multicast traffic. Unlike prior art methods, my method exploits the broadcasting nature of the downstream point-to-multipoint access system including EPONs. Operationally, the method transforms a multicast traffic shaping problem into a unicast traffic shaping problem by selecting a multicast subscriber and subsequently treating packets addressed to that selected multicast subscriber as unicast. Advantageously, these packets are still available to other members of the multicast group. And since these packets are treated as unicast from the perspective of traffic shaping, my inventive method significantly reduces traffic shaping complexity in multicast environment.

Abstract: A source specific multicast may be performed in a network, by inspecting a signal for a source specific multicast channel identifier. The source specific multicast identifier signal is then mapped to a frame switching identifier. The frame switching identifier can be mapped to the signal, allowing the signal to be directed a location based on the frame switching identifier.

Abstract: Channel changing can be accelerated by multicasting a bouquet of multicast burst streams from a server. In an example implementation, each multicast burst stream is delayed sufficiently so that a past independent frame is available for delivery to and display by a client device without waiting for a future independent frame. A multicast burst segment of a multicast burst stream includes a portion in which the bandwidth exceeds the nominal data rate of the underlying resource being streamed. The temporal delay between adjacent multicast burst streams in a multicast bouquet is set responsive to a maximum client delay time for tuning to a new resource stream.

Abstract: A switch for broadcasting packets is provided including a plurality of input ports, a switch fabric coupled to the input ports, a plurality of output ports coupled to the suited fabrics and a multicast interconnect coupled between the input ports and output ports for routing multicast packets directly between the input and output ports. The multicast interconnect may include a multicast queue, an arbiter coupled between the multicast queue and the input ports and a plurality of broadcast buffers coupled to the multicast queue, each broadcast buffer coupled to a corresponding output port. The multicast interconnect may include a plurality of egress arbiters coupled between corresponding broadcast buffers and output ports.

Abstract: Methods, systems, and articles of manufacture consistent with the present invention provide for a data processing system having an efficient network communication infrastructure. The data processing system has a plurality of data processing nodes distributed across a network. Each node comprises a processor and a memory having an application and a routing thread operatively configured to receive a message on the network. The message has an application identifier and a message group identifier. Each application has a command thread and a plurality of message group handlers. Each routing thread is operatively configured to determine whether the received message is for the application on the respective node and to notify the application's command thread of the received message in response to determining that the message is for the application. The command thread then directs the message to one of the message group handlers based on the received message's message group identifier.

Abstract: Method for generating and thereafter updating a data structure used for routing Internet protocol data packets. Routing a packet is performed by using a destination address of the packet and an updatable set of prefix rules. A prefix rule may be added to a first-level table if the terminating level of the prefix rule equals one. Otherwise, cascading tables may be created until reaching a terminating table for the prefix rule. Then, the prefix rule may be added to its terminating table. The data structure is updateable. The packet routing may be guided by associating one or more fields, or partial fields, of the most significant bits of a destination address of the packet with respective records of search tables, and using the last visited port identifier for routing there through the packet. The data structure is generated by a control processor and stored in a system memory, whereas a network processor searches in the data structure for a prefix rule suitable for each received packet.

Abstract: Active-active fibre channel capability in SAS and SATA apparatus and method is disclosed. In one embodiment, a storage device includes a dongle having a conversion module to translate between a fibre channel frame and a SATA frame and/or a SAS frame, and a data processing system to communicate through a fibre channel network with the storage device through two or more ports on the conversion module. An active-active module of the conversion module may provide multiple paths from the data processing system to the storage device. The conversion module may process the fibre channel frame and other fibre channel frames on a frame by frame basis. The dongle may be external to the storage device.

Abstract: An apparatus, data structures, and method are provided for prioritizing data transmissions within a network. As applied to a switching station in a network, the method prioritizes transmissions from the network to determine which packets should be transmitted from the switching station first when multiple packets are routed to the same outgoing port of the switching station. A packet prioritization station is provided, preferably as an add-on to the switching station. The packet prioritization station has a cache in which the destination address of each incoming packet is associated with every origin from which it has received a transmission within a certain time period. The packet prioritization station operates to give priority to transmissions to those destinations that have a higher number of associated origins. Thus, packets that are probably en route to time-critical users or groups of users will be sent before those that are less time-sensitive.

Abstract: A method of implementing association in an Automatic Switched Optical Network. When an initial node of the 1+1 protection service is outside the ASON, it associates the primary service and the slave service to realize 1+1 protection. This invention includes: a node of a first associated service acquiring path information of a second associated service from a node of the second associated service, and associating the second associated service to the first associated service according to the path information of the second associated service.

Abstract: In a base station of an IEEE 802.16 Wireless MAN-based portable Internet system, a forward packet scheduling method and apparatus is provided. The portable Internet system portable Internet service includes a user terminal AT for providing a portable Internet service, a base station AP for supporting an inter-user terminal wireless access and network connection, and a packet access router PAR for performing a mobility control and packet routing function of each user terminal AT. The forward packet scheduler includes a packet-based subsidiary scheduling part for considering a real-time service and a user-based scheduling part for ensuring fairness and increasing system efficiency (throughput). After finishing the 2-step scheduling, the generated frame data are transmitted though a physical layer to a wireless user terminal.

Abstract: A switching network for efficiently receiving and transmitting data packets having both frames and messages includes a crossbar switch with a plurality of surrounding ports for exclusively switching frames which normally consist of large data streams of 40 to 60 bytes. Then the ports are connected together in a message ring and small data entity messages, for example 4, 8, or 12 bytes, are switched from an input port to an output port around the ring avoiding congestion of the crossbar switch.

Abstract: A method and system provides narrowband communications across a communications link through the processing of a superframe of data into packets or through the processing of packets into a superframe of data. In an embodiment, a first node repackages a superframe of data containing multiple subframes of data in known positions within the superframe into multiple packets. A sequence indicator is inserted into a payload area of the multiple packets, the sequence indicator corresponding to a subframe in the given communications packet and its position within the superframe. The packets are transmitted across a connection to a second node. At the second node, sequence indicators in the payload portion of each of the packets are inspected. The multiple subframes of data are extracted along with corresponding command and control information. Using the sequence indicators, frames of data are formed from the multiple subframes of data.

Abstract: The invention concerns a data packet retransmission arrangement having a retransmission buffer, a counter, a forward error correction device and control logic. The retransmission buffer stores recently transmitted data packets. The counter keeps track of the number of retransmission requests received for a data packet (211). If this number is below a first integer value K, the data packet is retransmitted (212, 213). If this number is equal to or above the first integer value K, the forward error correction device calculates N forward error correction packets on L-1 recently transmitted data packets plus the data packet to be retransmitted (211), N being a second integer value equal to or larger than zero, and L being a third integer value equal to or larger than 1. In the latter case, the data packet is retransmitted together with the N forward error correction packets (214).

Abstract: A dual-mode broadband modem operative to operate in a service mode and a surveillance mode is disclosed. Generally, the dual-mode broadband modem comprises a primary communication device; a secondary communication device; a central processing unit in communication with the primary and secondary communication devices; and a memory unit in communication with the central processing unit. In the service mode, the primary communication device communicates with a broadband service provider and the central processing unit stores historical information regarding performance of the primary communication device in the memory unit. If an interruption in broadband service is detected, or the performance of the primary communication device falls below a predetermined threshold, the dual-mode broadband modem switches to surveillance mode and the secondary communication device is activated.

Abstract: A method for network services with a mobile vehicle comprising: initiating from a remote station a first connection session with a device in the mobile vehicle, wherein during the first connection session the device is notified of a pending command; and initiating a second connection session from the device in the mobile vehicle to the remote station in response to the first connection session, wherein the pending command is transmitted to the device in the mobile vehicle during the second session.

Abstract: A method and system for supporting an emergency call is disclosed. The method includes initiating (302) an emergency call through an Internet Protocol (IP) based system (108). The method further includes receiving (304) authorization for restricted access to the resources of the IP based system, only for the purpose of making the emergency call.