一种DTN模式,用于可靠的网络电话

C. Hoene, Patrick Schreiner
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引用次数: 1

摘要

众所周知,Internet只提供尽力而为的服务,这是IP电话的缺点。因此,不能保证最低的传输质量,特别是在网络拥塞或偶尔的链路故障时,基于udp的VoIP变得不可用。为了克服这些限制,我们开发了一种速率自适应传输系统,用于高度可扩展的语音和音频编解码器,该系统在非常低的比特率下使用延迟容忍网络(DTN)方法。这样,一旦可用的传输容量低于最低编码速率,我们的系统就切换到一键通(PTT)式的会话模式。主观会话质量测试表明,即使在传统的基于UDP的VoIP电话失败(MOS-CQS为1)的情况下,该算法也允许进行良好有效的会话(MOS-CQS为3.5)。
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A DTN mode for reliable internet telephony
IP telephony suffers from the well-known fact that the Internet does only provide a best-effort service. Thus, minimum transmission quality cannot be guaranteed and, especially during times of network congestion or occasional link failures, UDP-based VoIP becomes unusable. To overcome these limitations, we have developed a rate-adaptive transmission system for highly scalable speech and audio codecs that uses the Delay Tolerant Networking (DTN) approach for very low bit rates. In this way, as soon as the available transmission capacity falls below the minimum coding rate, our system switches to a Push-To-Talk (PTT) like conversational mode. Subjective conversational quality tests have shown that the algorithm allows for a good and effective conversation (MOS-CQS is 3.5) even in cases where traditional UDP based VoIP telephony fails (MOS-CQS is 1).
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