摘要:
A method to reduce the amount of bandwidth used in the transmission of digitized voice packets is described. The method is used to reduce the number of transmitted packets by suspending transmission during periods of silence or when only noise is present. The system determines if a background noise update is warranted based on human auditory perception factors instead of an artificial limiter on excessive silence insertion descriptor packets. The system searches for characteristics in the perceptual changes of background noise instead of analyzing speech for improved audio compression. The invention weighs factors affecting the perception of sound including frequency masking, temporal masking, loudness perception based on tone, and auditory perception differential based on tone.
摘要:
A method for constellation design in a telecommunications network (10) using pulse code modulation to transmit data signals upstream (30) between client and server voice-band modems (12,26). The invention selects a constellation for transmission over an analog channel of an equivalence class of data points using pulse code modulation based on the presence or absence of robbed bit signaling and interference from echo levels. The constellation is designed by determining noise in a PCM channel using a cumulative distribution function for echo (90), determining the extent of said noise for an array of possible constellation points (92), and selecting constellation points such that the largest negative noise of a first point remains above the largest positive noise level of a neighboring second point (94). Constellations are created off-line and stored (96) for retrieval during modulation depending on the level of the echo.
摘要:
A compensation system (figure 2A) for improving the accuracy of digital signals that are communicated through a digital network (113, figure 1) by reducing loss from attenuation quantization (DAQ) and rob bit signaling (RBS). The combined DAQ/RBS compensation system can be employed within the transmitting modem (101) connected to the digital network. A first adjustment mechanism combines a DAQ compensation quantity with each segment of the digital data. Next, the word is communicated to a linear-mu-linear converter (201), which is configured to simulate a digital transmission by mu-low encoding each digital data word into a code word and then subsequently mu-low decoding each code word back into linear digital data word. The linear-mu-linear converter includes an RBS compensation system (129) that causes an RBS compensation quantity to be mathematically combined with each segment to be tainted by RBS. A second adjustment mechanism combines the reciprocal of the DAQ compensation quantity with the linear digital data from the linear-mu-linear converter. Finally, the linear digital data word is passed from the linear-mu-linear converter to a linear-mu-converter (208) for conversion into a mu-law code word and transmission to the network.
摘要:
A telecommunications gateway allows packets to be sent over a TDM system and allows TDM traffic to be sent over a packet switched network. The gateway is a universal port that includes a plurality of Digital Signal Processors (DSPs) that are controlled by software. The controlling software determines what single function the DSP will perform over multiple channels. Each DSP handles multiple channels, however, each DSP is restricted such that all of its multiple channels are permitted to handle the telecommunications traffic according to one signaling protocol.
摘要:
A method of constructing a valid set of configuration parameters for ADSL2 and ADSL2+ compliant systems include selecting delays of a power of two ms, in which the ADSL2 or ADSL2+ compliant system may also possess a selected minimum noise protection value to produce maximum downstream and upstream net data rates corresponding to the selected delay and the selected minimum noise protection value.
摘要:
A compensation system (figure 2A) for improving the accuracy of digital signals that are communicated through a digital network (113, figure 1) by reducing loss from attenuation quantization (DAQ) and rob bit signaling (RBS). The combined DAQ/RBS compensation system can be employed within the transmitting modem (101) connected to the digital network. A first adjustment mechanism combines a DAQ compensation quantity with each segment of the digital data. Next, the word is communicated to a linear-mu-linear converter (201), which is configured to simulate a digital transmission by mu-low encoding each digital data word into a code word and then subsequently mu-low decoding each code word back into linear digital data word. The linear-mu-linear converter includes an RBS compensation system (129) that causes an RBS compensation quantity to be mathematically combined with each segment to be tainted by RBS. A second adjustment mechanism combines the reciprocal of the DAQ compensation quantity with the linear digital data from the linear-mu-linear converter. Finally, the linear digital data word is passed from the linear-mu-linear converter to a linear-mu-converter (208) for conversion into a mu-law code word and transmission to the network.
摘要:
A method of discriminating voice, data, and facsimile calls communicated through a voice-over-packet network. The gateway (2) is provided with software which can identify the existence of an answer signal (ANS) or a modified answer signal (ANSam) communicated between an answering modem (4) and an originating modem (1) over a packet network (5) during a voice state call. The originating gateway (2) can generate an ANS tone according to the protocols of the originating modem (1), using an originating-side gateway (2), when the existence of the ANS signal is identified by the receiving-side gateway (3). The originating gateway also generates an ANSam tone according to the protocols of the originating modem, using the originating-side gateway, when the existence of the ANSam signal is identified by the receiving-side gateway.
摘要:
A method for designing an equalizer (32) and tracking performance for upstream PCM in a digital communications network is described. The invention optimizes upstream data rates for data transmissions in a network between a client modem and a server modem. During training, the upstream channel impulse response (46) is identified compensating for any robbed bit signaling. The upstream transmit equalizer (32) is computed in closed form based on the identified channel (34). An equalizer in the receiver (41) is also used to track timing and channel variations. The invention approximates bit error rate performance by looking for code violations in the trellis code decoder and tracking their frequency. The bit error rate is used to determine if the current modem parameters need to be re-designed.
摘要:
A method of discriminating voice, data, and facsimile calls communicated through a voice-over-packet network. The gateway (2) is provided with software which can identify the existence of an answer signal (ANS) or a modified answer signal (ANSam) communicated between an answering modem (4) and an originating modem (1) over a packet network (5) during a voice state call. The originating gateway (2) can generate an ANS tone according to the protocols of the originating modem (1), using an originating-side gateway (2), when the existence of the ANS signal is identified by the receiving-side gateway (3). The originating gateway also generates an ANSam tone according to the protocols of the originating modem, using the originating-side gateway, when the existence of the ANSam signal is identified by the receiving-side gateway.