We have developed a "policy server for an access network (PSAN)" that can provide a policy-based IP virtual private network (IP-VPN) service. PSAN uses a "network management layer adaptation policy action table (NAPAT)" to provide a policy adaptation function, which converts from a user policy rule to certain policy rules at the network device level. Each network device level policy rule is suited to each unit of transport equipment of public networks, which employ a variety of network technologies (e.g., WDM, ATM, and SDH). To promptly provide the acceptance or rejection of the service level agreement (SLA), we use the "QoS capability network view (QCNV)" to instantaneously check if the user's QoS policy can be enforced. We developed a prototype system based on the asynchronous transfer mode passive optical network (ATM-PON) that we have already developed, and we showed that it can realize an IP-VPN service by an evaluation of scalability.

The scope of network management systems has been expanding rapidly, from simple network element management systems to integrated management systems that include network management and service management. Since the telecommunications world has become an open market, sales competition has intensified. Accordingly, a flexible, extendable, and reusable network management system is required. A distributed object-oriented environment meets these requirements, and CORBA (Common Object Request Broker Architecture) is a promising candidate as a distributed environment. An integrated management system should provide management capabilities that are independent from management protocols and from the location of Managed Objects (MOs) which represent managed resources. The system should also be able to execute multiple management operations (multiple management scenarios) concurrently. In this paper, we describe a technique of introducing an interface for each MO, called a Proxy MO, that encapsulates the management protocols and locations of the MOs and which are concurrently controlled in the CORBA environment. This technique provides a flexible, extendable, and reusable network management system.

A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed. It is based on time scale modification and sub-band coding with the aid of ADPCM with a multiquantizer. For time scale modification, the decimation/interpolation technique is introduced for the unvoiced period. Furthermore, in order to get computational accuracy of the pitch extraction for the voiced period, a method using the normalized autocovariance function is proposed. The algorithm was confirmed by computer simulations. The segmental SNR was about 13-16 dB for Japanese short sentences. A good mean opinion score value was also obtained by means of a subjective evaluation test.>

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding.>

This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance. The coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ) which selects the optimum ADPCM coder frame by frame and switches to it dynamically. To implement this coding algorithm, we used two Fujitsu DSPs (MB8764), 1.5 chips for the encoder and 0.5 chip for the decoder. Reconstructed speech with a 21 dB segmental SNR was obtained. With error correction, this codec provides good speech quality even with a bit-error rate of 10^-2 to 10^-3. To improve the subjective quality of the reconstructed speech, adaptive postfiltering was also applied. Since the processing delay of this codec is less than 10 ms, no echo-canceller is needed. Moreover, 2400 bps voice band data (CCITT Rec.V. 26) could be transmitted with a data error rate from 10^-7 to 5×10^-6, and G.III facsimiles were successfully transmitted using this codec.

Adaptive differential PCM (ADPCM) is an effective coding scheme to simplify the hardware and shorten the processing delay to realize a high-efficiency speech codec. The ADPCM with Multi-Quantizer (ADPCM-MQ) coding has been proposed as one of the highly efficient coding methods. In the ADPCM-MQ codec several ADPCM coding blocks with different quantization step-size update rates are operated in parallel, and the quantizer that gives the best characteristics is found and selected dynamically for each frame. This paper describes a new 8 to 9.6 kbps ADPCM-MQ coding algorithm that includes tree coding to improve the per-sample quantizing characteristic, and sub-band coding with high frequency band reconstruction to realize a lower bit rate coding. Computer simulation indicates good quality for speech reproduced by this algorithm with a segmental signal-to-noise ratio of 14 to 15 dB. Adaptive postfiltering can be added to enhance the subjective characteristics of the reproduced speech.