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LAN Based Low Delay Real Time Voice Communication System with Adaptive Jitter Control

機(jī)譯:具有自適應(yīng)抖動(dòng)控制的基于局域網(wǎng)的低延遲實(shí)時(shí)語(yǔ)音通信系統(tǒng)

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Progress being made in the development of multimedia network environments with net- worked multimedia PCs is leading to the day when LAN-based real time voice communication systems can be achieved. The major problems with LANs are delay, jitter and packet loss due to their best effort transfer characteristics; accordingly, the major issues of LAN-based systems are how to reduce end-to-end delay, absorb jitter and recover packet loss in order to provide high quality bi-directional real time voice communication. In this paper, a LAN-based real time voice communication system using IP protocols with adaptive jitter control is proposed. The advantageous characteristics of this system are the introduction of an adaptive jitter con- trol mechanism that minimizes end-to-end delay by optimizing jitter buffer size, the adoption of small algorithm delay hardware codecs, and the use of a TCP (UDP)/IP environment for voice communication in order to integrate data and voice applications. Results obtained in tests on a prototype system show that delay of less than 100 ms is achieved, which satisfies the ITU-T G.114 Recommendation of 150 ms as an asceptable range for bi-directional realtime voice communication. In addition, stable voice quality is ashieved that is little affected by the disturbance caused by dynamic changes in network load.
機(jī)譯:借助聯(lián)網(wǎng)的多媒體PC在多媒體網(wǎng)絡(luò)環(huán)境的開發(fā)中取得的進(jìn)展導(dǎo)致可以實(shí)現(xiàn)基于LAN的實(shí)時(shí)語(yǔ)音通信系統(tǒng)的時(shí)代。局域網(wǎng)的主要問題是由于盡力而為傳輸特性導(dǎo)致的延遲,抖動(dòng)和數(shù)據(jù)包丟失。因此,基于LAN的系統(tǒng)的主要問題是如何減少端到端的延遲,吸收抖動(dòng)并恢復(fù)丟包,以提供高質(zhì)量的雙向?qū)崟r(shí)語(yǔ)音通信。本文提出了一種具有自適應(yīng)抖動(dòng)控制功能的基于IP協(xié)議的基于LAN的實(shí)時(shí)語(yǔ)音通信系統(tǒng)。該系統(tǒng)的優(yōu)勢(shì)在于引入了自適應(yīng)抖動(dòng)控制機(jī)制,該機(jī)制通過優(yōu)化抖動(dòng)緩沖區(qū)大小來最小化端到端延遲,采用了小的算法延遲硬件編解碼器,以及使用了TCP(UDP)/ IP環(huán)境用于語(yǔ)音通信,以便集成數(shù)據(jù)和語(yǔ)音應(yīng)用程序。在原型系統(tǒng)的測(cè)試中獲得的結(jié)果表明,可以實(shí)現(xiàn)小于100 ms的延遲,這滿足了ITU-T G.114建議書中150 ms作為雙向?qū)崟r(shí)語(yǔ)音通信的可接受范圍。另外,實(shí)現(xiàn)了穩(wěn)定的語(yǔ)音質(zhì)量,幾乎不受網(wǎng)絡(luò)負(fù)載動(dòng)態(tài)變化引起的干擾的影響。

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