【作者】 杨明；

【导师】 杨宗凯； 杜旭；

【作者基本信息】 华中科技大学 ， 通信与信息系统， 2004， 硕士

【摘要】 VoIP是一种以IP电话为主,并推出相应增值业务的技术。其最大的优势是能广泛地采用全球IP互连的环境,提供比传统业务更多、更好的服务。采用VoIP技术的IP电话是以IP为标志的网络分组化和以流媒体传输为目标的网络业务综合化两大主流技术融合的结果,并成为传统电信与IP网络优势互补的一个突破口。然而由于IP分组网络自身的局限性,其在传输文字、图形等离散媒体时尚能胜任,而在传送语音、视频等实时媒体时则显得有点力不从心。如何在因特网上高质量地传输实时语音流媒体业务,已成为VoIP的关键问题之一。在这样的背景下,作者结合参与的VoIP接入网关的研发工作,对实时语音传输进行了深入的研究,从时延、抖动、丢包等出发对语音质量提出了改进的措施与算法,并在此基础上实现了多播音频会议。文章首先说明了实时语音传输的产生背景以及语音编码、信令技术、流媒体协议等相关技术。接着较为详细的介绍了实时传输协议RTP/RTCP,并对协议使用环境进行了分析。然后详尽的阐述了基于嵌入式Linux的实时语音传输设计方案,该方案采用了环形多缓冲技术和多线程技术实现了基于RTP的多播音频会议,并对多路混音提出了解决方案;通过去抖缓冲算法有效去除语音包的抖动问题,在时延与抖动之间达到良好平衡;并对语音质量密切相关的丢包、溢出、语音平滑等问题给予了改善。文章的最后对实时语音传输占用带宽、时延给出了测试结果,并对RTP在大规模应用中的可扩展性以及存在的问题提出了一些看法。更多还原

【Abstract】 VoIP is a technology that mostly utilized in IP phone, and relevant value-added services. The most advantage of VoIP is that can make use of the global IP internetworking environment, to provide the more and more、the better and better services than tradition PSTN network. The IP phone using VoIP technology is the result of the combination of such chief technology, the packet technology on network taking IP as its sign, and the integrating technique of services on network taking real-time media transmission as its goal. It has become the joint of traditional telecom and IP network, mutually supplying each other’s advantages.Due to the intrinsic character of IP network, it maybe be competent for transmission discrete media such as text、picture etc, but it is insufficient to transmit real-time media such as audio、video and so on. How to transmit real-time media with high performance over IP network that has become the most important problem in VoIP field.Under such a background, as a part of the research work on VoIP gateway, the author carried through the real-time voice transmission deeply, and bring forward some measures to improve voice quality on delay、jitter、packet lost aspects, then realize the multicast audio meeting.This article first introduced some background knowledge including voice encoder/decoder、signaling technology、stream media protocol. And then gave an exhaustive description on RTP/RTCP protocol, analyze the protocol using scenarios. Afterward provided a real-time voice transmission resolution upon embedded Linux. This resolution adopted multiple ring buffer and multi-thread technology to implement multicast audio meeting based on RTP/RTCP, supply resolved means on multiple voice synchronization; decrease the network voice jitter through the de-jitter buffer algorithm, <WP=6>and try to find the trade-off between the voice delay and jitter; At the last of article, the author provided the test results about real-time voice transmission bandwidth、delay, put forward some perspective views tailor to problems which exists on large-scale RTP/RTCP application scenery.更多还原