摘要
基于会话初始化协议(SIP)的VoIP系统在Internet上已经取得了广泛应用,但在目前的实际网络环境中,由于大量NAT设备的存在,使对等网络(P2P,Peer to Peer)之间的呼叫和数据通信难以实现。分析了四种NAT的类型特点,介绍了现有的NAT穿越方法,提出了一种基于STUN与TURN方式相结合的实现各种NAT穿越的VoIP系统设计方案。该方案对SIP信令采用可靠的TCP传输方式,对流媒体数据采取最大交付的UDP传输方式。经过校园网内部之间的网络环境测试,该方案达到了很好的接通率。
Session Initialization Protocol（SIP）-based VoIP systems have been widely applied over Internet recently.However,with the wide use of Network Address Translation（NAT）facilities,peer-to-peer（P2P）call becomes hard to realize in Intranet.This paper describes the mapping rules of four types of NAT,and analyzes the existing traversal methods,then proposes a VoIP system solution with the function of all types of NAT traversal requirements based on STUN and TURN methods.This design employs TCP protocol for SIP message transmission and UDP protocol for voice message transmission.Tests in campus environments show that this solution can achieve high performance.
出处
《通信技术》
2010年第11期105-107,共3页
Communications Technology
关键词
网络地址转换
SIP
P2P
STUN
TURN
Network Address Translation（NAT）
Session Initialization Protocol（SIP）
P2P
STUN
TRUN
