Abstract: A communication system and method usable to facilitate communication between a hearing user and an assisted user. In particular, the system employs a wireless portable tablet or other portable electronic computing device linked to a captioning enabled phone as a remote interface for that phone, thereby providing an assisted user with more options, more freedom, and improved usability of the system.

Abstract: Apparatus and methods for providing high-bandwidth, low-latency data service over a content delivery including existing wireline infrastructure. In one embodiment, a network architecture having service delivery over at least portions of extant hybrid fiber coax (HFC) infrastructure is disclosed, which includes standards-compliant ultra-low latency and high data rate services (e.g., 3GPP and IEEE Std. 802.11 services) via a common service provider. In one variant, an expanded frequency band (e.g., 1.6 GHz in total bandwidth) is used over the coaxial portions of the HFC infrastructure, which is allocated to two or more sub-bands. Wideband amplifier apparatus are used to support delivery of the sub-bands to CPE within the network.

Abstract: A method to present communications may include captioning, by a human assistant during a call between a first user using a first captioning telephone device and a second user using a second telephone device, words spoken by the second user into the second telephone device. The method may also include presenting the captioned words on a first display of the first captioning telephone device, receiving text typed into the second telephone device by the second user, and presenting the received text on the first display of the first captioning telephone device.

Abstract: Many mobile devices are used for documenting different scenarios that are encountered by the users as they go about their daily lives. In many situations, a mobile device may be used to document the scenario. This data may be of significant forensic interest to an investigator. In many situations, the owner of the phone may be willing to provide the investigator access to this data (through a documented consent agreement). Such consent is usually contingent upon the fact that not all the data available on the phone may be extracted for analysis, either due to privacy concerns or due to personal reasons. Courts have also opined in several cases that investigators must limit data extracted, so as to focus on only “relevant information” for the investigation at hand. Thus, only selective (or filtered) data should be extracted as per the consent available from the witness/victim (user). Described herein is the design and implementation of such a targeted data extraction system (TDES) for mobile devices.

Abstract: A password input and identification system includes a one-button password input component and a controller. The one-button password input component generates a series of trigger signals in response to a series of triggers that correspond to a series of user-input digits. The controller determines a value of each of the user-input digits by counting a number of the trigger signal(s) that correspond(s) to the user-input digit to serve as the value of the user-input digit. The controller compares the user-input digits with the predetermined password after the input of the user-input digits is finished, and performs a predetermined action when the user-input digits match the predetermined password.

Abstract: A method and system for authorizing a subject user to perform an action within an electronic service, receiving, a request to access the electronic service, the request including an indication of the subject user, of an object, and of the action that the subject user is desirous of performing on the object within the electronic service, based on at least one of the indication of the subject user and the object, retrieving, by the server, a set of predetermined rules each predetermined rule having been coded and stored, by the server, as a respective bytecode portion, executing the set of predetermined rules, analyzing an outcome rendered by the execution of each rule of the set of predetermined rules, and responsive to the outcome being indicative of a positive result, authorizing the subject to perform the action on the object.

Abstract: This application relates to methods and apparatus for configuring a serial data interface, especially a data interface (300) for a transducer (324) such as a digital microphone or loudspeaker. The data interface is selectively operable in a first or second serial data mode of operation for input of data to or output data from the transducer device. The first and second serial data modes are different, e.g. correspond to different serial formats. The data interface has a controller (330) configured to determine a resistance value (331, 332) at a sense terminal (314) of the transducer device and to control the data interface in the first serial data mode if the resistance value is within a first resistance range and control the data interface in the second serial data mode if the resistance value is within a second, different, resistance range.

Abstract: An embodiment of a graphics apparatus may include a frame divider to divide a frame into two or more sub-frames, and a parallelized post-render stage communicatively coupled to the frame divider to process a sub-frame of the two or more sub-frames in parallel with a render operation. The parallelized post-render stage may include a post-processor communicatively coupled to the frame divider to post-process a rendered sub-frame in parallel with the render operation. Other embodiments are disclosed and claimed.

Abstract: A host interface for a storage module may include an out-of-band (OOB) detector that is configured to detect receipt of an OOB signal using a clock signal. The clock signal may be generated by a clock generator that is activated using a counter. When an OOB signal is received, the counter may activate the clock generator. When no OOB signal is being received, the counter may wait for a predetermined time period before deactivating the clock generator.

Abstract: Aspects are disclosed for publishing contents of a content store. A storage operation is performed, and a completion of the storage operation is detected. Here, the storage operation redundantly stores contents of a content set onto instances associated with a content store. The contents stored in the instances are then published in response to the completion of the storage operation. In another aspect, a dataset table is generated to facilitate storing contents of a content set, which include payload and metadata. The payload is stored onto a payload table, and the metadata is stored onto a metadata table. For this embodiment, the dataset table includes a first foreign key to the payload table, whereas the metadata table includes a second foreign key to the dataset table. The dataset table is monitored to determine a storage status of the contents, and the contents are subsequently published based on the storage status.

Abstract: A method for maintaining the connection between a terminal and a network server is described. The terminal has a first processing unit with a first state and a second state, wherein the power consumption of the second state is less than the power consumption of the first state. When the first processing unit is in the second state, the terminal or the network server sends heartbeat data packets periodically in order to maintain the connection between said terminal and said network server. During the period that the terminal or the network server sends a periodic heartbeat data packet, a main processor of the terminal will not be woken up Thus, the terminal in a sleep state can maintain connection with the network server, and not be woken up due to the heartbeat data packets sent by the network server, which reduces the power consumption of the terminal.

Abstract: The disclosure discloses a telepresence method, a telepresence terminal and a telepresence system.

Abstract: Interdomain federation, generation or management of a virtual network (slice) that extends among domains can be performed without changing a function of a server or a node of a network domain without a function to manage the other domain. Pseudo node receiving a management message with the same protocol as a general node is added, in interdomain management information from an own domain, the other domain part in management information is received as the management information by the pseudo node (the other domain part is concealed) and when interdomain management information is transmitted to the other domain through the pseudo node by a federation protocol and the pseudo node and the interdomain management information is received from the other domain, the other domain part in the management information is bound as the management information of the pseudo node and transmitted to a management server of own domain.

Abstract: A system, method, and computer readable medium for remotely migrating from a first service to a second service, comprising sensing a ring wire state, sensing a tip wire state and switching a switch device to a direct current termination based upon a difference between the ring wire state and the tip wire state.

Abstract: Transmission parameters are determined for communication between apparatuses. These transmission parameters may specify, for example, a number of spatial streams, a bandwidth, and a transmission rate. Based on the determined parameters, a table of a set of tables comprising transmission parameters is identified and an index from the table is determined. Rate information is then sent in a frame from one of the apparatuses to the other. The rate information comprises several fields indicative of the index, the number of spatial streams, and bandwidth. The apparatus that receives the rate information may then use this information to identify the table and obtain transmission parameters from the table.

Abstract: According to certain aspects, the present invention provides methods and apparatuses for managing a low power mode in xDSL systems, and more particularly directed to a L2 mode exit procedure for VDSL systems that is robust and quick. In embodiments, parameters for exiting a low power mode are communicated between upstream and downstream modems before the low power mode is entered. According to certain aspects, these parameters include configurations for incrementally exiting low power mode in a plurality of stages. Embodiments of the invention include quickly estimating SNR at one or more stages of this plurality of stages. Additional or alternative embodiments include reliably signaling the beginning of low power mode exit. According to certain aspects, such signaling can include a synchro sequence of inverted and normal sync symbols.

Abstract: Methods and apparatus for voice and data interlacing in a system having a shared antenna. In one embodiment, a voice and data communication system has a shared antenna for transmitting and receiving information in time slots, wherein the antenna can only be used for transmit or receive at a given time. The system determines timing requirements for data transmission and reception and interrupts data transmission for transmission of speech in selected intervals while meeting the data transmission timing and throughput requirements. The speech can be manipulated to fit with the selected intervals, to preserve the intelligibility of the manipulated speech.

Abstract: In one embodiment, source data for a communication session may be split into an audio portion for transmission on a phone channel and a non-audio portion for transmission on a data channel. A server and a phone may accordingly establish an audio portion of a communication session on the phone channel. In response to a trigger, the server may provide a push notification on the data channel to the phone, where the push notification is associated with an application executing on the phone that is configured to participate in the non-audio portion of the communication session on the data channel with the server. Upon obtaining the push notification on the data channel during the audio portion on the phone channel, the application may correspondingly activate on the phone to participate in the non-audio portion of the communication session during the phone's participation in the audio portion (e.g., merging the portions).

Abstract: Prediction of a channel capacity is accomplished based on a TDR echo without explicitly estimating the topology of the line. The prediction is based on obtaining a measured TDR echo, determining a theoretical TDR echo for a plurality of loop lengths, estimating the equivalent TDR length based on an optimization, updating the equivalent TDR length and utilizing the updated TDR length to predict one or more of the upstream and downstream data rates.

Abstract: One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset.

Abstract: Novel tools and techniques are provided for delivering plain old telephone service (“POTS”) telephony over high speed data networks. In particular, various embodiments provide tools and techniques for concurrent transmission of POTS voice signals and data signals over the same wire(s) of high-speed data lines or data cables. Various systems and methods might, in some instances, utilize upbanding or rebanding of the POTS voice band to a higher frequency band above the data stream band spectrum for transport of voice concurrent with data over the same wire(s) in the cable. The system might comprise interface devices at either end of a cable segment, one interface device to reband the voice signal and to combine the voice signal with the data signal for each dual-transport wire in the cable, and another interface device at the other end to separate the voice signal from the data signal.

Abstract: A method and arrangement are disclosed in a Vector Control Entity for enabling a fair bit rate distribution among lines of similar priority in a vectoring group, when applying partial vectoring in a DSL communication system. The method comprises calculating a rate balancing metric, RBM, for each line i within a priority group A, indicative of the ratio between the bit rate of a line i with a current vectoring resource allocation, and the estimated bit rate of line i assuming approximately no crosstalk within the vectoring group. The method further comprises allocating partial-vectoring resources to the line/s within the group A, based on the calculated RBMs, such as to reduce the difference in RBM between the lines. The method further comprises calculating an updated RBM for any line within the group A subjected to changes in vectoring resource allocation. The method may be iterated until certain criteria are fulfilled. The arrangement is adapted to enable the performance of the above described method.

Abstract: A service outlet for coupling a data unit to a wired digital data signal and for coupling a service unit to an analog service signal, for use with a service wire pair installed in walls of a building, the service wire pair concurrently carrying a wired bi-directional digital data signal and an analog service signal carried over a service signal frequency band, using frequency division multiplexing, wherein the wired digital data signal is carried over a frequency band distinct from the service signal frequency band. The outlet has a single enclosure and, within the enclosure: a wiring connector; first and second filters coupled to the wiring connector; a service connector coupled to the first filter and connectable to the service unit for coupling the service unit to the analog service signal; a service wiring modem coupled to the second filter; and a power supply coupled to the service wiring modem.

Abstract: A method and a device adjust a power allocation of users in a digital subscriber line environment. An intermediate power allocation is determined for at least one user initializing with the digital subscriber line environment based on a new power allocation determined for the digital subscriber line environment containing the at least one user. The intermediate power allocation provides a predefined minimum signal-to-noise ratio margin for the active users of the digital subscriber line environment. Furthermore, a communication system can contain such a device.

Abstract: A method and corresponding device for determining a data connection profile for a data connection (15) between one of a plurality of end-user devices (14, 16, 18) and an aggregation transceiver device (24, 34), the data connection belonging to one of a plurality of predetermined categories, each category being associated with one of a plurality of service providing entities (62, 64, 66), the method comprising receiving performance data indicative of the performance of a data connection; receiving an entity identifier indicative of one of said plurality of service providing entities, and determining in dependence thereon the category to which the data connection belongs; and determining a data connection profile for the data connection in dependence on the performance data and on the category so-determined, the data connection profile specifying a set of values for one or more parameters associated with the data connection.

Abstract: A method and a device adjust a power allocation of users in a digital subscriber line environment. An intermediate power allocation is determined for at least one user initializing with the digital subscriber line environment based on a new power allocation determined for the digital subscriber line environment containing the at least one user. The intermediate power allocation provides a predefined minimum signal-to-noise ratio margin for the active users of the digital subscriber line environment. Furthermore, a communication system can contain such a device.

Abstract: Network apparatuses, networks, computer program products, and management station operational methods are provided. One aspect of the invention provides a network apparatus including a management station adapted to couple with a network including a plurality of managed devices, the management station being configured to output a plurality of initial commands for application to respective managed devices, the initial commands being configured to stimulate initial responses from the managed devices, the management station being further configured to receive the initial responses, to identify responding ones of the managed devices responsive to the received initial responses, and to provide an asset table containing the identified managed devices.

Abstract: A method for minimizing message collision in a device is presented. Two or more overlapping real-time streaming simplex audio messages are received. One of the audio messages is forwarded to be reproduced while the other is stored. Forwarding of the delayed audio message is delayed such that overlapping reproduction of the audio messages is minimized. Reproduction is delayed until a predetermined clock time expires or is dependent on one or more of: the length of the second audio message or the amount of overlap of the first and second audio messages.

Abstract: A real-time bandwidth monitor (RTBM) for VoIP applications are disclosed having a media streaming function to sense the available bandwidth between two endpoints of a VoIP communication (herein, a “call path”) utilizing the media streaming function and to adapt in real-time the transmission rate of the media stream to utilize that bandwidth. The media stream may include voice content, video content, or other media content.

Abstract: The present invention is directed to a method for transmitting data over a network. The method includes ascertaining at a first node the need to transmit one or more data streams from the first node to a second node over the network, the one or more data streams including time-based media that is generated either locally or remotely on the network. The method also includes ascertaining at the first node a timeliness preference for reviewing the one or more data streams by one or more recipients respectively. The ascertained timeliness preference for each one of the data streams is either (a) time-sensitive where at least one recipient of the one data stream has indicated an intent to review the one data stream in a near real-time mode as received by the recipient over the network while the time-based media of the one data stream is being generated or (b) less-than time-sensitive.

Abstract: A device for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band), signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components.

Abstract: A video and digital audio transport system transports high resolution computer, video, computer video graphics and other video signals, as well as multiple channels of high quality audio, over extended lengths of unshielded twisted pair (UTP) cabling. Frequency compensation techniques are employed to extend the bandwidth of the UTP cables, and differential signaling over the common mode of a first two of the four twisted pairs comprising the UTP cable is used to transmit horizontal sync and differential signaling over the common mode of a second two of the four twisted pairs is used to transmit vertical sync. In addition, a unique digital encoding technique is applied to a standard digital audio encoding protocol to augment the number of available audio channels. A digitally controlled delay line is incorporated to compensate for the different lengths of each twisted pair.

Abstract: In one embodiment, a server may obtain source data for a communication session having a combined audio portion and non-audio portion. As such, the server may actively split the audio portion of the communication session from the non-audio portion of the communication session, and then provide the audio portion (to a first set of one or more phones) over a phone channel and the non-audio portion (to a second set of one or more phones) over a separate data channel. The second set of phones may then obtain and merge the audio portion from the phone channel and the non-audio portion from the data channel to reestablish the full communication session.

Abstract: A third-party interactive voice response service includes a media archiving service in which media streams from a call to a subscriber is encrypted by a public key of the subscriber. The media streams may optionally be compressed and are encrypted in real-time so that at any time no tangible portion of the media streams can be accessed by the third party provider. The multiple media streams can optionally be compressed and/or encrypted individually or after they have been combined into a combined stream. The subscriber is able to retrieve the encrypted media streams and decrypt them using the subscriber's private key. The encryption and compression are under program control of the interactive voice response service.

Abstract: A method to increase spectral efficiency in a communication system is described herein. The communication system includes at least one mobile station and is capable of transmitting messages encoded according to a plurality of available modulation coding schemes (“MCSs”). Each available MCS includes a modulation scheme and an effective coding rate. The MCSs are indexed according to increasing complexity. A signal-to-interference ratio (“SINR”) is determined which is sufficient to satisfy a predetermined frame error rate (“FER”). A first MCS and a corresponding amount of transmissions needed to satisfy the predetermined FER at the SINR using the first MCS are determined. The first MCS has a higher effective coding rate than a second MCS. The second MCS sufficiently satisfies the predetermined FER at the SINR in a single transmission. A message encoded according to the first MCS is transmitted through the communication system using hybrid automatic repeat request (“HARQ”).

Abstract: A method and a device adjust a power allocation of users in a digital subscriber line environment. An intermediate power allocation is determined for at least one user initializing with the digital subscriber line environment based on a new power allocation determined for the digital subscriber line environment containing the at least one user. The intermediate power allocation provides a predefined minimum signal-to-noise ratio margin for the active users of the digital subscriber line environment. Furthermore, a communication system can contain such a device.

Abstract: A device for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band), signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components.

Abstract: A method and arrangement are disclosed in a Vector Control Entity for enabling a fair bit rate distribution among lines of similar priority in a vectoring group, when applying partial vectoring in a DSL communication system. The method comprises calculating a rate balancing metric, RBM, for each line i within a priority group A, indicative of the ratio between the bit rate of a line i with a current vectoring resource allocation, and the estimated bit rate of line i assuming approximately no crosstalk within the vectoring group. The method further comprises allocating partial-vectoring resources to the line/s within the group A, based on the calculated RBMs, such as to reduce the difference in RBM between the lines. The method further comprises calculating an updated RBM for any line within the group A subjected to changes in vectoring resource allocation. The method may be iterated until certain criteria are fulfilled. The arrangement is adapted to enable the performance of the above described method.

Abstract: In one embodiment, source data for a communication session may be split into an audio portion for transmission on a phone channel and a non-audio portion for transmission on a data channel. A server and a phone may accordingly establish an audio portion of a communication session on the phone channel. In response to a trigger, the server may provide a push notification on the data channel to the phone, where the push notification is associated with an application executing on the phone that is configured to participate in the non-audio portion of the communication session on the data channel with the server. Upon obtaining the push notification on the data channel during the audio portion on the phone channel, the application may correspondingly activate on the phone to participate in the non-audio portion of the communication session during the phone's participation in the audio portion (e.g., merging the portions).

Abstract: Precise operation of an accelerator operating member (7) is more difficult to perform in a reversing operation of a vehicle (1) as compared with an advancing operation. Therefore, behavior of the vehicle (1) is not smooth and thus tends to be unnatural. In the reversing operation of the vehicle (1), driving force output from an internal combustion engine (2) is limited in accordance with vehicle acceleration (D). At this time, it is possible to regulate the driving force of the vehicle (1) in conformity to the actual operation of the accelerator operating member (7) by the driver. In addition, limitation of the driving force is not executed in the advancing operation of the vehicle (1). At this time, the vehicle (1) can be driven in a state in which the driving force is comparatively small. Therefore, it is possible to prevent the behavior of the vehicle (1) from being unnatural.

Abstract: This relates to a method for determining a receive bit rate over a data link. The method includes setting a receive bit rate to an initial bit rate. A bit pattern over the data link is detected using the initial bit rate. A determination is made as to whether the detected bit pattern matches one of a plurality of known bit patterns corresponding to fractional bit rates. If the detected bit pattern does not match one of the plurality of known bit patterns, then a determination is made as to whether the received data includes a framing error. If it does, then the receive bit rate is lowered to a low detection bit rate that is lower than the fractional bit rates corresponding to the plurality of known bit patterns.

Abstract: A network for transporting power and multiplexed data and digital telephone signals. The network includes at least three nodes and first and second wiring segments in a building for carrying the multiplexed data and digital telephone signals, and at least one of the segments is configured to additionally carry a power signal. A power consuming component is connected to the at least one wiring segment and is powered by the power signal carried by that segment. Each wiring segment connects a different pair of the nodes together to form, with nodes nodes, a packet based bi-directional communication link. One of the nodes contains communication link composed of a repeater, a bridge, or a router connectable to a data unit. At least one of the nodes is connected to a remote data unit external to the building for coupling the remote data unit to at least one of said communication links.

Abstract: The telecommunication system for communication between PSTN and analog FAXes and/or analog phones by using separated passes for data and synchronization clock flow.

Abstract: A service outlet for coupling a data unit to a wired digital data signal and for coupling a service unit to an analog service signal, for use with a service wire pair installed in walls of a building, the service wire pair concurrently carrying a wired bi-directional digital data signal and an analog service signal carried over a service signal frequency band, using frequency division multiplexing, wherein the wired digital data signal is carried over a frequency band distinct from the service signal frequency band. The outlet has a single enclosure and, within the enclosure: a wiring connector; first and second filters coupled to the wiring connector; a service connector coupled to the first filter and connectable to the service unit for coupling the service unit to the analog service signal; a service wiring modem coupled to the second filter; and a power supply coupled to the service wiring modem.

Abstract: A method for minimizing message collision in a device is presented. Two or more overlapping real-time streaming simplex audio messages are received. One of the audio messages is forwarded to be reproduced while the other is stored. Forwarding of the delayed audio message is delayed such that overlapping reproduction of the audio messages is minimized. Reproduction is delayed until a predetermined clock time expires or is dependent on one or more of: the length of the second audio message or the amount of overlap of the first and second audio messages.

Abstract: Systems and methods of Voice over Internet Protocol (VOIP) communication are provided. A particular system includes a VoIP endpoint. The VoIP endpoint is adapted to monitor a packet loss measure between the VoIP endpoint and a second VoIP endpoint during a VoIP communication using a first CODEC. The VoIP endpoint is also adapted to negotiate with the second VoIP endpoint to select a substitute CODEC having a nominal data rate that is lower than an in-use nominal data rate of the first CODEC.

Abstract: A method for reliably transferring media over a network from a sending node to a receiving node. The method includes ascertaining at the sending node if the media to be transmitted is either time-sensitive or not time-sensitive. For media ascertained as non time-sensitive, the media is transmitted by adjusting the rate of transmission at the sending node based on network conditions. As the non time-sensitive media is received, the receiving node generates one or more low priority requests for the retransmission of any missing non time-sensitive media lost during the transmission. The method also includes transmitting the time-sensitive media from the sending node to the receiving node. As the time-sensitive media is received, the receiving node ascertains if a predetermined acceptable network transmission loss level is met. If met, the receiving node generates one or more low priority requests for the retransmission of any missing time-sensitive media lost during transmission.

Abstract: Communication apparatus having interfaces for exchanging data with first and second neighbors, a memory for storing codec information regarding the communication apparatus and a control entity operative to detect a message from the first neighbor, the first message being indicative of codec information regarding an originating entity. In response, the control entity assesses compatibility between the codec information regarding the originating entity and the codec information regarding the communication apparatus. If the assessment is positive, the control entity self-identifies the communication apparatus as a candidate for terminally supporting a subsequent codec-bypass negotiation with the originating entity. If the assessment is negative, the control entity self-identifies the communication apparatus as a candidate for non-terminally supporting such negotiation.

Abstract: A system for the end-to-end delivery of digital television signals. In a preferred embodiment a digital television signal is: received from production equipment, typically in HD format at approximately 1.4 gigabits per second (Gbps); the received signal is transmitted to a venue point-of-presence; converted for transmission via a 270 Mbps local loop; transmitted to a fiber network point of presence/video service edge; packetized into TCP/IP packets in a video gateway; and routed to one or more destination addresses via the fiber network; received at one or more video service edge destinations; converted to a digital television format, typically SDI; and either transmitted via a second 270 Mbps local loop for delivery to a customer site and subsequent conversion to a 1.4 Gbps HD signal, or converted directly to a 1.4 Gbs HD signal at the receiving video service edge.

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.