Research On VoiceXML-Based Voice Value-Add Service Platform And Its Algoritms

Voice services will be the most important in next generation network as wellas they are in the current communication networks. Voice services hold the largestpercentage of the total revenues, and are the main profit source of the networkoperators. The sharp growth of voice value-added services (WAS) does not onlyprofit the operators directly, but also improves the utilization ratio of the presentequipments, brings the customers new voice experiences, increases their loyaltiesto the operators, and brings more potential customers and profits to the otherservices of the operators.But there are many shortcomings in the present voice value-added serviceplatforms (VSP), such as closed, not flexible, hard to maintain and deploy the newservices, etc. With the growth of the WAS and the user amount, the presentVSPs could not meet the demands, and began to cumber the further developmentof the WAS. So, it is time for the emergence of new VSP. VoiceXML (Voiceextensible Markup Language), which has been specified for the voice browser,has the virtue of flexibility, easily develop, and is one of the best choices of VSP.This thesis is jointly supported by National Science Fund for DistinguishedYoung Scholars (No. 60525110) , Program for New Century Excellent Talents inUniversity (No. NCET-04-0111) , Specialized Research Fund for the DoctoralProgram of Higher Education (No. 20030013006) to research the voicevalue-added platform based on VoiceXML. A voice value-added service systembased on the research result had been developed, which has been further deployedin several provinces of China and is serving millions of users.The principal contributions of the work presented in this thesis are:1) VoiceXML-based VSP gets the VoiceXML service scripts and resourcesvia the Internet, so the network delay is unavoidable, while the phone users are much sensitive to the delay. The prefetch schema in the VoiceXML-based VSPhas been studied to resolve this problem, and it is proposed that the prefetchedobjects should be the resources which are referenced by the VoiceXML scripts.An adaptive multi-user shared Markov predict algorithm is presented, which usesthe character that the voice platform can know whether a user is online or not.This algorithm can predict the probability of the forthcoming required resourcesof all the online users, which is helpful to improve the veracity of the prediction.And a preemptive priority model is designed to schedule the prefetch tasks, whichmaps the resource access probability to the task priority. The simulation researchshows that the precision of the adaptive multi-user shared Markov predictalgorithm is better than that of single user Markov predict algorithm and thepreemptive priority schedule model can get more profits than the round-robinschedule model. The combination of the two new algorithms can considerablyreduce the network delay, accelerate the response, and decrease the user's waitingtime.2) In order to further reduce the network delay, the cache replacementalgorithm has been studied. By analyzing the present cache replacementalgorithms, two key problems are pointed out, one is how to establish the resourceutility function, and the other is the algorithm of replacement process. For the firstproblem, the LRU-K (K-Least Recently Used) algorithm is improved byproducing a novel utility function named PLRU-K (Perfect LRU-K). And for thesecond problem, according to the 0/1 Knapsack problem, the 1-optimal greedyreplacement process (1-GRP) is proposed to select and replace resources from thecache. The simulation research shows that the utility function of PLRU-K can getthe more veracious future profit of resource than that of LRU-K and P-LFU(Perfect Least Frequently Used) and 1-GRP algorithm can get more profit thanP-GRP (Profit-based Greedy Replacement Process) and PD-GRP(Profit-Density-based Greedy Replacement Process). So, the performance of thecache replacement algorithm based on PLRU-K and 1-GRP goes beyond otheralgorithms, especially when the cache volume is far smaller than the total size ofthe resources.3) The cache consistency control algorithm is indispensable in order to avoidgiving some stale data to the user. It is discussed how to develop a method that does not entail any server modifications or changes to the HTTP (HyperTextTransfer Protocol) to implement the consistency control. By analyzing the presentconsistency algorithms, considering the character of the VoiceXML document,and according to the conclusion that the Web pages modification intervals followexponential distribution, the Fitting & Prediction algorithm is proposed. Itestimates the validity of the cached document by performing parameter fits tostochastic distribution and predicting the change probability of VoiceXMLdocument. Simulation research indicates the algorithm surpasses the Alexprotocol and can obtain a stale ratio lower than 0．01% to meet the demand of thevoice platform while effectively enhance the system performance.4) To resolve the problem that the multi-party communication function isinvalid in VoiceXML, it is proposed that the function can be performed byextending the object element. After the discussion of the function details of themulti-party communication, two object extended schemes included block andnon-block approaches are proposed and their advantage and disadvantage arepointed out. And it is illustrated how to use those objects.5) Audio mixing is an essential component in conference, and the presentaudio mixing algorithms have a protean volume. By analyzing those algorithms,an algorithm named asymmetrical wave-shrinking is proposed. Based on the factthat the low volume appears more frequently than the high volume in the voicesignal, it uses a fixed mixing weight independent of inputs to mix the audios.Without multiplication and division operations, the algorithm is so simple and fastthat it can be easily implemented by hardware. The experimentation shows thatthe result of this algorithm is very good and sounds naturally and fluently withoutnoises even if there are many inputs. This algorithm is one of the fastestalgorithms at present, so it can meet the demand of the high performance and highconcurrence in the voice conference.The contributions in this thesis are not only used in VoiceXML-based voicevalue-added platform, and are also useful for the other voice platforms and theother related research fields.