Abstract: Integrating external on-demand services is performed through embedding a pseudo on-demand video-title for each new on-demand external service, which is activated by an original session manager (SM) once a particular set-top converter box (STB), which belongs to a particular group_id, initiates a request for the external service. An STB application intercepts SM tuning information on the selected transport_id and service_id of the pseudo service and recovers the address of the selected edge device and QAM in the original system, associated with the intercepted transport_id, through the inverse mapping the couple (group_id, transport_id) to the edge and QAM IP and UDP port addresses which map is set by external management system.

Abstract: Apparatus and methods are provided for controlling synchronous and asynchronous communications in an avionic network. The method includes receiving one or more asynchronous messages and one or more synchronous messages, allocating each of the synchronous messages to a corresponding predetermined time slot while producing an unoccupied portion in at least one of the predetermined time slots, and allocating the asynchronous messages to one or more of the unoccupied portions of the predetermined time slots.

Abstract: The present disclosure discloses a method for transmitting an IP message, negotiating a bandwidth saving capability and saving network bandwidth. The method for transmitting an IP message includes: sending more than one bandwidth saving capability supported by a sender to a receiver; receiving one bandwidth saving capability selected by the receiver; obtaining a type of an IP message for transmitting data according to the bandwidth saving capability selected by the receiver; and sending the IP message to the receiver after constructing the IP message, wherein the one bandwidth saving capability is selected from the more than one bandwidth saving capability and is supported by the receiver. By applying the methods provided by the present disclosure, the bandwidth and resources of an IP bearer network in a communication system are saved.

Abstract: In one embodiment, the present invention is a method and apparatus for scalable virtual private network multicasting. In one embodiment a service network builds a new data multicast distribution tree for each high-bandwidth multicast data flow (e.g., multicast data flows that require an amount bandwidth meeting or exceeding a predefined threshold). However, if the multicast data flow is a low-bandwidth flow (e.g., if the required amount of bandwidth falls below the predefined threshold), the multicast data flow is routed over an existing multicast distribution tree in order to minimize an amount of state information that must be maintained by service provider core routers in the backbone network.

Abstract: A communication server implements a method that minimizes bandwidth usage between a media device and the communication server in a communication system where separate session and floor control protocols are used. Where multiple clients have established separate sessions to access a media source at a media device and upon determining that the multiple clients are attempting to access the same media source, the communication server implements floor control logic to tie multiple floor arbiters together across the separate sessions to cause a single media stream to be sent from the media source and received at the communication server for distribution to the clients, thereby minimizing the bandwidth on the uplink between the media device and the communication server.

Abstract: A method and apparatus encode a source data stream via convolutional encoding or selected encoding scheme. Plural encoded data streams are interleaved and transmitted on a transmission channel. Data groups generated via convolutional or selected encoding are interleaved via time-interleaving functions to disperse selected bits within puncture groups of the data groups, bits in between data groups, and bits in selected sets of data groups to facilitate reconstruction of the source data stream from at least a portion of the interleaved data stream received on at least one transmission channel. The time-interleaving functions are selected to facilitate reconstruction of the source data stream from one transmission channel following continuous blockage. Subsets of bits of puncture groups are selected to allow reconstruction of the source data stream from more than one of plural transmission channels using a minimum number of subsets.

Abstract: According to the invention, an information output, e.g. a voice output, video information output, or a simultaneous output of video and audio information, which is to be transmitted via a packet-oriented network, is generation by means of pre-coded information output components. Said pre-coded information output components are present in a memory system and are retrieved as part of the processing of a request for the output of information by an information output system. Similarly, generation rules for the information output can be stored in the memory system and accessed by the information output system in order to generate the information output. According to one embodiment, information output components or generation rules are created in an information output configuration system and are subsequently made available in the memory system. The invention optimizes the use of resources for information-output services.

Abstract: Messages transmitted between a receiver and a transmitter are used to maximize a communication data rate. In particular, a multicarrier modulation system uses messages that are sent from the receiver to the transmitter to exchange one or more sets of optimized communication parameters. The transmitter then stores these communication parameters and when transmitting to that particular receiver, the transmitter utilizes the stored parameters in an effort to maximize the data rate to that receiver. Likewise, when the receiver receives packets from that particular transmitter, the receiver can utilize the stored communication parameters for reception.

Abstract: Aspects of a method and system for reducing transceiver power via a variable number of channels are provided. In this regard, utilization and/or availability of network and/or device resources may be determined and a configuration of channels utilized for transmitting data may be determined based on the determined utilization and/or availability of resources. Accordingly, the number of channels over which data is communicated may be altered based on determined thresholds for the utilization and/or availability of resources. Moreover, the configuration of channels utilized for communicating data may be dynamically altered by monitoring changes to the utilization and/or availability of resources. For example, the number of channels may be altered based on available bandwidth on one or more channels, based on capacity and/or available space of one or more buffers, and/or based on available power or desired power consumption of a transmitter.

Abstract: Load balancing for layer-2 link aggregation involves initial assignment of link aggregation keys (LAGKs) and reassignment of LAGKs when a load imbalance condition that merits action is discovered. Load conditions change dynamically and for this reason load balancing tends to also be dynamic. Load balancing is preferably performed when it is necessary. Thus an imbalance condition that triggers load balancing is preferably limited to conditions such as when there is frame drop, loss of synchronization or physical link capacity exceeded.

Abstract: Methods and apparatus for performing multiplexing of video or other content (e.g., programs) within a network using feed-back from a subsequent digital program insertion stage, and/or feed-forward information from a prior multiplexing stage. In one embodiment, the network comprises a hybrid fiber coax (HFC) cable network having headend and hub-based statistical multiplexing stages, and communication between the two stages is used to improve the visual quality performance and bandwidth utilization of the output multi-program stream during conditions where downstream content is inserted into the transport stream. Business methods associated with the various multiplexing features described above are also disclosed.

Abstract: A medium access control (MAC) layer architecture and functionality for supporting enhanced uplink (EU). A MAC entity for EU, (i.e., a MAC-e entity), is incorporated into a wireless transmit/receive unit (WTRU), a Node-B and a radio network controller (RNC). The WTRU MAC-e handles hybrid-automatic repeat request (H-ARQ) transmissions and retransmissions, priority handling, MAC-e multiplexing, and transport format combination (TFC) selection. The Node-B MAC-e entity handles H-ARQ transmissions and retransmissions, E-DCH scheduling and MAC-e de-multiplexing. The RNC MAC-e entity provides in-sequence delivery and handles combining of data from different Node-Bs.

Abstract: A system for verifying a cipher-based message authentication code (CMAC), including a reception (RX) module logically residing between a physical layer controller (PHY) and a media access controller (MAC) processor, such that the RX module is configured to receive one or more portions of the CMAC with one or more bursts, process the one or more bursts, and write the one or more portions of the CMAC to one or more memory locations in a memory. The system also includes a transmission (TX) module logically residing between the PHY and the MAC processor, such that the TX module configured to verify the CMAC concurrently as the RX module processes the one or more bursts.

Abstract: Addressing methods and apparatus which use airlink resources in an efficient manner are described. In order to use airlink resources efficiently, a variety of address types are supported. The number of bits in the address may vary depending on the type. In order to support a variety of different address types, of different lengths, in one embodiment, an address used for communications over an airlink is constructed by including in the address an address type field used to communicate an address type indicator and, optionally, an address field used to communicate an address value. The address field is variable in length and may be null, requiring the communication of no bits in this field, for certain types of addresses. Four types of addresses which may be supported include: i) an Access Terminal assisted address, ii) a Network assisted address, 3) a Pilot PN based address; and 4) a reserved address.

Abstract: In establishing a data connection with a wireless wide area network, a wireless device may evaluate one or more factors in order to select between IP header compression methods to be used for traffic in the data connection between the wireless device and an access point for the wireless wide area network. For example, the wireless device may select the IP header compression method based on a remaining battery level of the wireless device, a type of application on the wireless device initiating the data connection or another factor. In one embodiment, the selection of the IP header compression method may also be based on an amount of available bandwidth for the access point to allocate to wireless devices.

Abstract: More customization and adaptation of coded data streams may be achieved by processing the information signal such that the various syntax structures obtained by pre-coding the information signal are placed into logical data packets, each of which being associated with a specific data packet type of a predetermined set of data packet types, and by defining a predetermined order of data packet types within one access unit of data packets. The consecutive access units in the data stream may, for example, correspond to different time portions of the information signal. By defining the predetermined order among the data packet types it is possible, at decoder's side, to detect the borders between successive access units even when removable data packets are removed from the data stream on the way from the data stream source to the decoder without incorporation of any hints into the reminder of the data stream.

Abstract: More customization and adaptation of coded data streams may be achieved by processing the information signal such that the various syntax structures obtained by pre-coding the information signal are placed into logical data packets, each of which being associated with a specific data packet type of a predetermined set of data packet types, and by defining a predetermined order of data packet types within one access unit of data packets. The consecutive access units in the data stream may, for example, correspond to different time portions of the information signal. By defining the predetermined order among the data packet types it is possible, at decoder's side, to detect the borders between successive access units even when removable data packets are removed from the data stream on the way from the data stream source to the decoder without incorporation of any hints into the reminder of the data stream.

Abstract: More customization and adaptation of coded data streams may be achieved by processing the information signal such that the various syntax structures obtained by pre-coding the information signal are placed into logical data packets, each of which being associated with a specific data packet type of a predetermined set of data packet types, and by defining a predetermined order of data packet types within one access unit of data packets. The consecutive access units in the data stream may, for example, correspond to different time portions of the information signal. By defining the predetermined order among the data packet types it is possible, at decoder's side, to detect the borders between successive access units even when removable data packets are removed from the data stream on the way from the data stream source to the decoder without incorporation of any hints into the reminder of the data stream.

Abstract: Thus, the exemplary embodiments of the invention describe methods, computer program products and apparatus that provide improved quantization, as may be useful within the context of a communication system (e.g., a wireless communication system) that has a relay node. In one non-limiting, exemplary embodiment, a method includes: receiving a transmission having source data from an information source; determining an estimate of the source data using a quantization technique based on maximizing data throughput; and transmitting a message including the determined estimate towards an information destination.

Abstract: A transmitter for transmitting data packets including a Media Access Control (MAC) address extractor configured to extract one or more MAC addresses from packet headers included in data packets, a MAC compression unit configured to provide compressed MAC data including one or more pointers associated with each of the one or more extracted MAC addresses respectively, a packet modifier configured to modify the data packets by adding the compressed MAC data to the data packets, and a transmission unit configured to transmit the modified data packets. A receiver for receiving data packets including compressed MAC data, the compressed MAC data including pointers associated with MAC addresses, including a receiving unit configured to receive data packets, a compressed MAC data extractor configured to find and extract compressed MAC data from received data packets, and a packet modifier configured to modify the received data packets by removing the compressed MAC data from the received data packet.

Abstract: A method for transmitting a header-compressed packet based on a retransmission mechanism is disclosed according to the present disclosure. The method mainly includes: a header compressor of a sending device obtains feedback information of a header-compressed packet in a transmission process; and the header compressor determines a state of the header compressor according to the feedback information, compresses a header for the packet according to the state of the header compressor, and sends the packet to a decompressor of a receiving device. According to the present disclosure, a header compression mechanism is combined with an automatic retransmission mechanism, so that the header compressor obtains feedback information of an automatic retransmission technology, and estimates the decompression information of the decompressor correctly, so as to change a state machine of the header compressor at proper time and improve transmission efficiency of the packet.

Abstract: An architecture that enables distributed storage capacity to be shared as though accesses were being made to a Fiber Channel Switch is described. The distributed capacity may include Fiber Channel devices and thus a method for addressing Fiber Channel devices coupled in a resilient ring has been provided. Expected Fiber Channel behavior and reliability are attained in the architecture of this invention through the application of bandwidth provisioning and extended reach mechanisms at each node in the ring. The bandwidth provisioning mechanisms ensure that access is provided to the communication links of the ring for each node according to a defined bandwidth distribution. The extended reach mechanisms of the invention ensure that the standard Fiber Channel packet delivery assumptions are satisfied. The solution exploits native ring characteristics to deliver customer separation and management of the resulting storage network.

Abstract: A method and a system for setting up header compression communication and a header compression policy function entity are disclosed herein. The method includes: a header compression enforcement function entity receives a header compression indication from a header compression policy function entity; and the header compression enforcement function entity negotiates header compression channel parameters with the other corresponding header compression enforcement function entity to set up header compression channel. A communication system and a header compression policy function entity are disclosed. The embodiments of the present disclosure overcome the failure of implementing the header compression communication in the prior art.

Abstract: A method for compressing a user datagram protocol (UDP) data packet includes: when receiving an Internet Protocol version 4 (IPv4) UDP data packet, a compressor searching a context of the UDP data packet, if the context does not exist, creating the context of the UDP data packet, taking a value of an IP identifier (IP-ID) field of the UDP data packet as an initial value of a serial number (SN) of the context, and then compressing the UDP data packet according to the context. Also provided are methods and devices for compressing and decompressing a UDP data packet. The methods and devices can effectively improve the compression efficiency, reduce the probability of sending the IR/IRDYN packet, significantly reduce 1 byte transmission, improve the compression efficiency by 50%, and reduce the probability of using an extension header with a high capacity.

Abstract: A method for transmitting and receiving data between a base station and a terminal in an E-UMTS (Evolved Universal Mobile Telecommunications System) or an LTE (Long Term Evolution) system is disclosed. A header overhead of data transmitted from a wireless end can be minimized by adding a header field existence indicator indicating presence or absence of a header in an upper data block to a data block of a lower layer, to thus effectively use radio resources.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: The described embodiments include a system that modulates the width of a high-speed link. The system includes a transmitter circuit coupled to a high-speed link that includes N serial lanes. During operation, while using a first number of lanes to transmit frames on the high-speed link, the transmitter circuit determines a second number of lanes to be used to transmit frames on the high-speed link based on a bandwidth demand on the high-speed link. The transmitter circuit then sends an indicator of the second number of lanes to a receiver on the high-speed link. Upon receiving an error-free acknowledgment of the indicator from the receiver, starting from a predetermined frame, the transmitter circuit transmits subsequent frames on the high-speed link using the second number of lanes.

Abstract: A method and apparatus for operation of a node within a mobile ad hoc cognitive radio network is provided. The method includes sensing at least one assigned communications channel. Sensing at least one assigned communications channel includes measuring a value of at least one parameter corresponding to the communications channel. The method further includes comparing the measured value of the at least one parameter with a set of stored values of the at least one parameter to determine a change in the measured values. Finally a sleep mode of the node is activated for a time period, wherein the time period is determined using the change in the measured values.

Abstract: A system and method for partitioning a bandwidth of a single channel among plural multimedia streams in a time varying manner. The partitioning is undertaken by dynamically establishing first and second bit rates respectively associated with first and second multimedia streams.

Abstract: An apparatus and method for combining transponder bandwidths comprises a wide-band virtual transponder for transmitting a single data stream. The wide-band virtual transponder is comprised of a plurality of narrow-band physical transponders. A plurality of elementary streams are statistically multiplexed to create the single data stream, wherein the single data stream is forward error correction encoded and demultiplexed into a plurality of transponder streams for transmission by the plurality of physical transponders. The physical transponders each use a different portion of a signal spectrum, wherein the different portion may be guard bands or a combination of legacy bands and guard bands. Upon receipt, the transponder streams are multiplexed to recover the single data stream, wherein the recovered single data stream is forward error correction decoded and statistically demultiplexed to recover the plurality of elementary streams.

Abstract: A method and apparatus for requesting bandwidth in a subscriber station is disclosed. The method dynamically changes the size of the bandwidth request based on the prediction of the number of packets needed to be transmitted. The average delay experienced by packets in a queue is measured, and this information is then used to change the coefficient of a prediction equation. When the experienced average delay is below the agreed upon QoS latency parameter or delay target, the method reduces the size of the bandwidth requests by making the prediction equation more conservative. On the other hand, when the experienced delay is above the agreed upon latency, the algorithm will make the prediction equation more aggressive, increasing the bandwidth requests and reducing the latency for future packets. By modifying the prediction equation based on the measured delay, the method is able to select the optimal point for achieving time delay requirements while preserving air-link resources.

Abstract: The wireless base station notifies the missing to the transmitter upon detection of missing of a data segment received from the transmitter, and the transmitter controls the header compression state of the data segment addressed to the wireless terminal upon the reception of the notification of the missing.

Abstract: A system involves a plurality of RF-enabled occupancy detectors. Each occupancy detector communicates with and controls an associated plurality of RF-enabled fluorescent lamp starter units. A network master has an RF transceiver used to communicate with the occupancy detectors using a first protocol, thereby retrieving status information from the starter units. The network master also has a second RF transceiver for communicating directly with a cellular telephone using a second protocol. A user can use the cellular telephone to control and interact with the lighting system through the network master, and/or to retrieve status information from the network master. The network master automatically generates and sends email alerts to the user by sending the alerts to an email server. The email server forwards the emails to the cellular telephone via a cellular telephone network. Alerts may, for example, indicate a low battery voltage condition or that a lamp needs replacement.

Abstract: A node of a telecommunications network comprises a first function configured to perform a first operation on a first portion of a packet handled by the node and a second function configured to perform a second operation on a second portion of the packet. The first function and the second function are configured to employ a shared transaction for operating on the packet whereby, by virtue of the shared transaction, after performance of the first operation and the second operation, the packet has less overhead attributable to the first function and the second function than if the shared transaction had not been employed in performance of the first operation and the second operation.

Abstract: An apparatus for controlling utilization of RTCP bandwidth for compound and non-compound RTCP packets is described. This apparatus includes a bandwidth detector determining available RTCP bandwidth and a channel quality detector determining channel quality. A bandwidth divider connected to the bandwidth detector and the channel quality detector divides the available RTCP bandwidth between compound and non-compound RTCP packets based on the determined channel quality.

Abstract: In one embodiment, whether packet loss of link layer packets at a receiver has exceeded a lost threshold is detected. Feedback is sent to a transmitter indicating that a header compressor, for compressing higher layer packets at the transmitter, should send a less compressed header if the packet loss of the link layer packets has exceeded the lost threshold. The higher layer packets are at a higher layer than the link layer and are formed by the link layer packets.

Abstract: A system and method for updating a display unit remotely located from a computing system are described. The method includes copying display information from a video buffer of a computing device executing an application to generate a display update, calculating a size of the display update, and determining an amount of available bandwidth of a network connecting the computing device and the display unit. The method also includes calculating a time period required to communicate the display update from the computing device to the display unit, transmitting the display update to the display; and repeating the method after the time period has elapsed.

Abstract: Systems and methods of exchanging voice chat information are provided. Voice chat information is received and the received voice chat information is placed into one or more packets. A control channel is selected and the one or more packets are transmitted over the selected control channel. The voice chat information can be transmitted in short messaging service (SMS) packets over the control channel. Moreover, voice chat information is exchanged between first and second communication stations by way of a SMS gateway, and media and signaling information are exchanged between the first and second communication stations independent on the SMS gateway.

Abstract: In a computing system, a method and system for a thin client and blade architecture are provided. A blade may generate video, audio, and peripheral control information that may be transmitted to a thin client (TC) by utilizing a video encoder, an audio bridge, and a peripheral bridge. Communication between the blade and the TC may occur based on a communication protocol that may operate independently of an operating system and/or applications running on the blade. The video encoder may dynamically compress the video information according to network capacity and/or video content and may dynamically select from various compression algorithms. The blade may configure and manage operations that interface with the TC. The TC may comprise a video decoder, a transceiver, a processor, a video display bridge, an audio bridge, and a peripheral bridge and may be adapted to communicate with peripheral devices.

Abstract: Provided are an apparatus for a dynamic header compression and a method thereof. The method for a dynamic header compression includes: receiving a received packet having a header from an upper layer; comparing header information of the received packet with header information of a previous packet; when a suppression part of the received packet is the same as a suppression part of the previous packet, compressing the received packet based on a predetermined header suppression rule; and when the suppression part of the received packet is not the same as the suppression part of the previous packet, transmitting the received packet without compression.

Abstract: Technique for transmitting Ethernet packets from an Ethernet source point to an Ethernet destination point via a communication link, comprising compression of the packets in such a manner that the source point transmits complete Ethernet packets, the communication link carries shortened information blocks (SIB) obtained from said complete Ethernet packets, while the destination point receives complete Ethernet packets essentially identical to those transmitted from the source point.

Abstract: A data transmission apparatus for sequentially transmitting data in units of packets each containing transmission data to the receiving end. The apparatus including a reception unit receiving the transmission data as an input signal, a packet formation unit receiving the transmission data, and forming an uncompressed packet in which predetermined transmission data is stored as uncompressed data, and a compressed packet in which at least a portion of transmission data that follows the predetermined transmission data is compressed and stored as compressed data. The apparatus also including a reference information management unit holding and managing as reference information related to the uncompressed packet, and a transmission unit transmitting the packets formed by the packet formation unit. The packet formation unit forming compressed data to be stored in a compressed packet, based on the transmission data of the uncompressed packet and the reference information.

Abstract: A method for transmission, a method for reception, a system, a transmitter and a receiver for signalling a hierarchical priority mapping of a transport stream (TS) to a receiver is disclosed. A way to separate tuning information between different TSs with different priorities. In a wireless hierarchical broadband transmission, preferably based on DVB-T, one signal may carry two transport streams: low priority (LP) stream and high priority (HP) stream, and both should be identified with their own respective transport_stream_id. One bit flag is added into a certain descriptor of the broadband transmission to indicate the hierarchy of the transport stream on which said descriptor is providing information. By the appliance of this priority indication or priority flag, the receiver can obtain hierarchy mapping of each transport stream announced in the administrative information of the broadband transmission.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: More customization and adaptation of coded data streams may be achieved by processing the information signal such that the various syntax structures obtained by pre-coding the information signal are placed into logical data packets, each of which being associated with a specific data packet type of a predetermined set of data packet types, and by defining a predetermined order of data packet types within one access unit of data packets. The consecutive access units in the data stream may, for example, correspond to different time portions of the information signal. By defining the predetermined order among the data packet types it is possible, at decoder's side, to detect the borders between successive access units even when removable data packets are removed from the data stream on the way from the data stream source to the decoder without incorporation of any hints into the reminder of the data stream.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, obtaining block information indicating a block type, and lossless decoding the first and second channel data based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: A method and apparatus provide reserved bandwidth and QOS/COS virtual circuit connections in a network using both conventional and novel reservation protocols and frame formats. An apparatus according to the invention includes an enterprise control point that communicates with switches via a reserved signaling channel. The switches have been upgraded or replaced to include enhanced functionality. The enhanced switches detect packets that include requests for reserved connections according to existing reservation protocols such as RSVP and IEEE 802.1P/Q. Such detected packets are forwarded to the enterprise control point for processing via a reserved signaling channel. The enterprise control point identifies a path within the network that can satisfy the requested QOS/COS and reserves the requested resources all along the path from beginning to end.