话音无阻塞的可视电话系统设计与实现  

Design and implementation for video phone system of non-blocking voice

作　　者：石硕[1] 李久仲[1] 郭庚麒[2] 李洛[1] 陈建国 

机构地区：[1]广东轻工职业技术学院计算机系,广州510300 [2]广东交通职业技术学院计算机系,广州510650 [3]广州加晋信息技术有限公司,广州510300

出　　处：《数字通信》2009年第4期63-67,共5页Digital Communications and Networks

基　　金：广东省科技计划项目(2007B080701005)

摘　　要：为实现话音无阻塞、视频图像清晰流畅和低延迟的高品质可视通话,提出了一种使用PSTN单独传送话音,使用IP网络单独传送通话者视频并在终端同步还原的可视电话系统及其实现方法,系统由终端和支撑服务器群构成。设计用PSTN的话音拨号信号或来电振铃信号作为视频呼叫的触发信号,终端通过捕获电话号码封装为SIP的Invite消息并发起视频呼叫,与对端建立P2P连接并使用UDP报文传送RTP视频流来实现视频通话。系统测试表明通话话音始终无阻塞;在带宽为256kbit/s、IP网络无拥塞的条件下,实现了视频CIF分辨率,帧频30fps,视频延迟小于60ms;用户使用习惯如电话号码和拨号方式无需任何改变。其综合性能优于现有的PSTN可视电话和IP可视电话。In order to achieve high-quality video calls with non-blocking voice, clear, smooth and low-latency video images, a video phone system which integrates the advantages of PSTN and IP network and its implementation method was proposed. The system consists of terminals and supporting servers. Separately, the voice is transmitted over PSTN and the video image is sent over Internet and they are restored simultaneously in the terminal. The terminal captures and put the dial-up signaling or ringing signaling package as the invite message of SIP for peer-to-peer connection of terminals. And the system uses UDP packets to send the RTP video stream. With non-blocking voice, video delay is less than 60ms under the conditions of bandwidth at 256Kbps, IP network without congestion, caller Videos for CIF resolution and its frame rate is 30 fps. Moreover, the terminal phone number and the mode of dial-up are the same as PSTN. Its comprehensive performance is better than PSTN video phone or IP video phone.

关 键 词：可视电话 二网融合 无阻塞 低延迟 

分 类 号：TN949.28[电子电信—信号与信息处理] TN927.2[电子电信—信息与通信工程]