Abstract: A system and method for estimating medical billing codes and patient's financial responsibility for the services availed or to be availed by patients from medical services providers and legal healthcare organizations is provided. The system provides a platform to users such as patients, medical services providers, and legal healthcare organizations to be informed in advance about estimated prices of medical services and medical insurance coverage to be availed by the patients. Patient's financial responsibility is calculated from medical concepts stored in the system and corresponding historical billing codes and medical insurance coverage for a patient. Further, methods of setting custom rules in the system to refine results based on patient demographics and other parameters are provided. Advantageously, the system provides automatic sharing of information among users and notifications on updated information and helps the users to enquire, network and market their services.

Abstract: A method and apparatus which provides that, in a voice over network, incoming packets are analyzed and the appropriate moment to increase or decrease the nominal delay associated with a jitter buffer is determined. Hence, the nominal delay is adjusted at an appropriate moment based on network jitter characteristics. Preferably, the nominal delay is adjusted when voice activity is absent. The method and apparatus provide for improved play out of the jitter buffer, and provide improved performance.

Abstract: In contrast to prior are solutions that conduct an averaging operation to estimate metrics related to a communications channel such as a signal-to-noise ratio, the present invention selects a particular value such as a minimum, maximum, or median value from a distribution of values collected over a selected interval. Selecting a particular value from the distribution of values facilitates a more accurate characterization process and increased data throughput. To reduce the processing burden associated with selecting a particular value, the present invention provides a set of cascaded value selection queues that each selects a particular value from the queued values such as a minimum value. The cascaded queues are also successively sub-sampled to reduce the computing resources required to characterize the communications channel. The estimated metrics resulting from the above-described process may be used to adjust the data encoding process and increase data throughput on the communications channel.

Abstract: A method and apparatus for using and combining sub-frame processing and adaptive jitter-buffers for improved voice quality in voice-over-packet networks. Data is placed in a jitter buffer, where the data has a frame-length consisting of a plurality of samples. Some of the samples are placed in the DMA buffer, and some of the samples are placed in the back-up buffer. Samples are read out of the DMA buffer, and samples are moved from the back-up buffer to the DMA buffer.

Abstract: In contrast to prior are solutions that conduct an averaging operation to estimate metrics related to a communications channel such as a signal-to-noise ratio, the present invention selects a particular value such as a minimum, maximum, or median value from a distribution of values collected over a selected interval. Selecting a particular value from the distribution of values facilitates a more accurate characterization process and increased data throughput. To reduce the processing burden associated with selecting a particular value, the present invention provides a set of cascaded value selection queues that each selects a particular value from the queued values such as a minimum value. The cascaded queues are also successively sub-sampled to reduce the computing resources required to characterize the communications channel. The estimated metrics resulting from the above-described process may be used to adjust the data encoding process and increase data throughput on the communications channel.

Abstract: Noise and signal-to-noise ratio (SNR) estimation are relatively straightforward tasks. However, when SNR is small, systematic errors in measurement may result in over-estimation of SNR, which also occurs during runtime monitoring of SNR. Here, sufficient numbers of bits have been preassigned to each channel using QAM modulation scheme. Therefore, SNR relative to QAM lattice size depends on the noise margin and the desired (bit error rate) BER. If a relatively small margin is desired, similar measurement errors may result in over-estimation of SNR. Another problem that arises is that the variance of the noise estimator is relatively high. Therefore, SNR estimates may vary by several dB, and there is only 50% confidence in the usual estimators that the actual SNR value will not be worse than that estimated. Thus, a computationally efficient method for SNR estimation that also allows for specification of a confidence level in the estimates is provided.

Abstract: A method and apparatus which provides that, in a voice over network, incoming packets are analyzed and the appropriate moment to increase or decrease the nominal delay associated with a jitter buffer is determined. Hence, the nominal delay is adjusted at an appropriate moment based on network jitter characteristics. Preferably, the nominal delay is adjusted when voice activity is absent. The method and apparatus provide for improved play out of the jitter buffer, and provide improved performance.

Abstract: A method and apparatus for using and combining sub-frame processing and adaptive jitter-buffers for improved voice quality in voice-over-packet networks. Data is placed in a jitter buffer, where the data has a frame-length consisting of a plurality of samples. Some of the samples are placed in the DMA buffer, and some of the samples are placed in the back-up buffer. Samples are read out of the DMA buffer, and samples are moved from the back-up buffer to the DMA buffer.

Abstract: A jitter generation apparatus and method configured to capture time varying aspects of jitter and packet loss in packet networks for testing/debugging jitter control algorithms. The jitter generation algorithm includes a probabilistic markov model which includes at least one hidden state and a plurality of state-dependent distributions. At each packet arrival, the probabilistic markov model emits a number corresponding to the delay for that packet that comes from a distribution which is unique to the state that the model is in. The model also includes a state transition matrix, and is configured to possibly make a transition to some other state, as defined by the state transition probability matrix, upon emitting the number. In order to model packet loss, a state may be provided which always results in infinite delay. The parameters of the model may be “trained” using actual, collected network data.