科研成果详情

VoIP应用程序可以以相对低廉的价格,为用户提供优质的语音甚至是视频实时通信服务.然而,IETF提供的请求/应答机制在当前充满NAT的网络环境面前,经常无法正常工作.目前的解决方案都是基于STUN协议的,而该协议会周期性地发出心跳消息以维持公网和私网地址的映射关系;这些消息对VoIP服务器来说是一种极大的资源浪费.设计并实现了一种基于SIP的自适应端到端通信系统,支持NAT下的端到端VoIP通信.系统扩展了SDP请求/应答模型,使其能够令一个会话中的各个端用户交换公网和私网地址/端口号.通过实现一个实际系统证明了该扩展方案的可行性.一系列实验证明了提出的系统相比其它基于超级结点转发机制的VoIP应用在效率和表现上的优越性. 

In recent years,VoIP applications are growing rapidly.However,the commonly used offer/answer mechanism cannot work properly in the environment where network address translators(NAT) are massively deployed.Current solutions rely on STUN,in which heartbeat messages would periodically be sent out to maintain the mapping between the public and private addresses.The state of maintaining heartbeat messages would become a huge burden to the VoIP server.We design and implement an adaptive SIP-based peer-to-peer communication system that supports VoIP over NATs.The system extends the SDP offer/answer model to share the public and private IP addresses and ports of both media ends at the call establishment of a session.We demonstrate its feasibility through the implementation of a real system.We evaluate the effectiveness of our proposed scheme and show that our system has a better performance when compared to other VoIP applications that use the supernode-relay mechanism.