Abstract: An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.

Abstract: A method and apparatus for routing voice packets through a communications network wherein the routing priority of the packets is based on a location of a packet within a sequence of transmitted voice packets, relative to one or more identified voice packets in the transmitted sequence. In one embodiment of the invention, the routing priority of a voice packet is based on the location of the packet in a sequence of packets relative to voice activity, such as the beginning, middle or end of a talk spurt. For example, the routing priority of the first several voice packets of a talk spurt is increased to compensate for the fact that there is less network jitter protection available at the receiver for these packets. In another embodiment, the routing priority of a voice packet may depend upon its location relative to a cellular handoff from one transmitter to another.

Abstract: A method and apparatus for defending against a Denial of Service attack wherein a target victim of an attack has recognized the existence of an attack and identified its source. The carrier network which provides service to the victim automatically receives one or more IP (Internet Protocol) source/destination IP address pairs from the victim, and then limits (e.g., blocks) the transmission of packets from the identified source address to the identified destination address. The carrier may implement this filtering capability as a stand-alone box included in the network, or as a line card incorporated into otherwise conventional network elements already present in the network. The source/destination address pairs to be blocked may be advantageously communicated from the victim with use of security signatures and with use of redundant connections from the victim to the carrier network to ensure receipt even under congested network conditions.

Abstract: A method and apparatus for enhancing voice intelligibility for network communications of speech such as, for example, VoIP (Voice-Over-Internet-Protocol), in the presence of packets which arrive too late for normal playout. When a late speech packet is received by a speech decoder, that packet and, if necessary, one or more additional packets subsequent thereto, are played out over a shorter than normal duration so that the decoder can “catch up” with the encoder. Since a voice frame is usually decoded in several sub-frames—typically two or three—this shortened playout may be achieved, for example, by skipping one sub-frame from each frame to be shortened.

Abstract: A method and apparatus for performing multiple descriptive source coding in which a plurality of homogeneous encoders are advantageously employed in combination with a corresponding plurality of advantageously substantially identical decoders. In particular, diversity is provided to the multiple encoders by modifying the quantization process in at least one of the encoders such that the modified quantization process is based at least on a quantization error resulting from the quantization process of another one of the encoders. In this manner, diversity among the multiple bit streams is obtained, and in particular, the quality of a reconstructed signal based on a combination of multiple decoded bit streams at the receiver is advantageously superior to that based on any one of the decoded bit streams. In accordance with a first illustrative embodiment of the present invention, two Pulse Code Modulation (PCM) coders are employed.

Abstract: A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.

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: A method for performing recoverable image and video watermarking which survives the use of block-based image and video compression techniques. One or more of the lowest order bits of the first DCT coefficient (the “DC” coefficient) which is to be coded are used as a “data channel” by which information representing a recoverable watermark may be embedded into an image or into a video signal frame. Encoding is performed by replacing one or more low order bits of the luminance value of each pixel in a block with a number of bits of the watermark data, and decoding is performed by averaging one or more low order bits of the decoded luminance values of the pixels in a block to retrieve a corresponding number of bits of the watermark data.

Abstract: The concept of “roaming” is extended to wireline networks with a network hosted technique which emulates a wireline telecommunications network user's environment when the user is using a wireline telecommunications device located at a physical location other than his or her “home” location, or when using a wireline or wireless telecommunications device other than his or her own regardless of the location. Any telephone or other telecommunications device being used by the user advantageously appears to be logically connected directly to his or her home telecommunications network, irrespective of the location of the telephone or other telecommunications device being used and the telecommunications network to which that device is actually connected.

Abstract: A method and apparatus for grooming traffic demands according to mileage based tariffs. An Integer Linear Program (ILP) that captures the traffic grooming problem is defined, and such a linear program can in principle be solved by conventional linear program application systems which are fully familiar to those of ordinary skill in the art. However, the time required to solve such an ILP is fairly large, even for the moderately sized networks we are interested in. That is, there are many possible routes to consider, and hence many integer variables in the ILP. Therefore, further in accordance with the principles of the present invention, the ILP is advantageously run on the Delaunay Triangulation of the network rather than on the completely connected network graph.

Abstract: A method and apparatus for performing Quality-of-Service (QoS) calculations on packet-based communications networks using a QoS measure which is based on data included in non-lost packets, as well as on data included in lost packets, when the proper interpretation of the data in non-lost packets depends upon data in one of the lost packets. Two new QoS measures that address the limitations inherent in the prior art PLR (Packet Loss Rate) measure are introduced. The Packet Loss Distortion Rate (PLDR) measure determines both packets which are lost, as well as packets whose proper interpretation depends on one or more packets which have been lost. The Media Distortion Rate (MDR) measures the actual quantity of media material that is lost, regardless of how the material is grouped into individual packets.

Abstract: An RF antenna grid design in which the antenna array is installed on a patch panel cover allows for the retrofitting of an RF antenna grid on existing patch panel based systems without disrupting network operation. An apparatus for determining connectivity between device ports on a patch panel and cable ends having corresponding RFID tags attached thereto comprises a patch panel having a plurality of device ports and a patch panel cover comprising a plurality of RFID antennas, where each of the RFID antennas comprises one or more protruding portions thereof, each of the protruding portions of the RFID antennas being in close physical proximity to one or more of the plurality of device ports when the patch panel cover is positioned in a fixed positional relationship to the patch panel (e.g., when the patch panel cover is “attached” to the patch panel).

Abstract: A method and apparatus for estimating transmission delay across a telecommunications network by performing a statistical analysis of conversational, behavior in the network. Certain characteristic events associated with conversational behavior (such as, for example, alternative silence events, double-talk events, talk-spurt events and pause in isolation events) are identified and measured. Then, based on the proportion of time that these events occur, an estimate of the delay is calculated using a predetermined equation. Illustratively, the equation is a linear regression equation which has been determined experimentally.

Abstract: A low complexity packet loss concealment method for use in voice-over-IP speech transmission calculates a cross-correlation of previous speech data to estimate the pitch period of the previous speech when speech frames have been lost. A tap interval used to calculate the cross-correlation is dynamically adapted, thereby reducing the computational complexity of the process. In addition, the pitch period estimation is bypassed completely when it is determined not to be necessary, as a result of the speech being unvoiced or silence. A waveform “bending” operation is performed into the current frame without inserting any algorithmic delay into each frame.

Abstract: A method and apparatus for performing multiple descriptive source coding in which a plurality of homogeneous encoders are advantageously employed in combination with a corresponding plurality of advantageously substantially identical decoders. In particular, diversity is provided to the multiple encoders by modifying the quantization process in at least one of the encoders such that the modified quantization process is based at least on a quantization error resulting from the quantization process of another one of the encoders. In this manner, diversity among the multiple bit streams is obtained, and in particular, the quality of a reconstructed signal based on a combination of multiple decoded bit streams at the receiver is advantageously superior to that based on any one of the decoded bit streams. In accordance with a first illustrative embodiment of the present invention, two Pulse Code Modulation (PCM) coders are employed.

Abstract: Private certificates designed to counteract problems associated with certificate lending are configured such that disclosure of a secret key associated with one certificate automatically results in disclosure of a secret key associated with another certificate, while the corresponding public keys are unlinkable with one another. In an illustrative private certificate generation protocol, a user generates verification information associated with a first public key. The verification information is generated at least in part using a corresponding first secret key. The verification information is supplied to a certification authority, which generates based at least in part on the first public key and the verification information a second public key having a corresponding second secret key, and generates a certificate based at least in part on the second public key. The private certificate generation protocol may be asymmetric or symmetric.

Abstract: A Distributed Denial-of-Service (DDoS) attack by a TCP stateless hog is defeated with use of an enhancement to the keep-alive mechanism provided by RFC 1122. A TCP server receives a new TCP connection request from a possible attacker and sends a keep-alive probe packet back thereto using an “invalid” sequence number. Illustratively, this “invalid” sequence number comprises a random number selected to be reasonably distant from the actual current sequence number. When a responsive packet is received from the potential attacker, the TCP server verifies the accuracy of the acknowledgement number in the received packet, thereby determining whether the potential attacker may be a TCP stateless hog.

Abstract: A method and apparatus for measuring Quality-of-Service (QoS) in packet-based communications networks for Voice-over-IP applications is advantageously based not only on packet loss information but also on particular characteristics of the speech data itself. Measurement of failures of the “quasi-stationary” (QS) property of speech—that is, the assumption that both the pitch and spectral envelop usually vary slowly relative to the packet size—are advantageously used in conjunction with measures of packet loss characteristics (e.g., packet loss rates or burst ratios) to calculate a QoS measure. One specific QS failure measure—referred to herein as QSFE (quasi-stationary failure estimate)—advantageously estimates the phoneme rate and is used in one illustrative method for calculating a QoS measure for Voice-over-IP applications in the presence of packet loss.

Abstract: A method and apparatus for detecting previous packet loss in non-packetized speech by applying one or more filters to a segment of said non-packetized speech, each of said one or more filters determining an energy parameter value for a given frequency band of said segment of said non-packetized speech; comparing one or more of said determined energy parameter values to one or more corresponding thresholds; and detecting previous packet loss based on said comparison of said one or more of said determined energy parameter values to said one or more of said corresponding thresholds.

Abstract: A method and apparatus for scheduling the transmission of data packets over a multihop wireless backhaul network in which delay guarantees through the network may be advantageously ensured. Illustratively, a novel packet scheduling scheme is provided which is advantageously based on an existing scheduling policy for wireline networks and for which a delay guarantee (based on the delay characteristics of the existing scheduling policy) can be advantageously ensured. In one embodiment, an even-odd link activation framework is defined for a given multihop wireless backhaul network, and an associated scheduling policy based on an arbitrary existing scheduling policy for wireline networks is derived and adopted. In such a case, the derived scheduling policy, when applied to the given multihop wireless backhaul network, advantageously ensures a worst-case delay guarantee of approximately twice that of the existing scheduling policy for wireline networks.