人在旅途--<<A JAIN-SIP Applet Phone>>学习笔记

人在旅途发表于 2006/3/2 10:10:53

　这个程序和InstantMessaging不同,它可以用来观察IP电话从拨打(invite)到挂机(bye)的一系列SIP消息. 1.安装从http://snad.ncsl.nist.gov/proj/iptel/下载该软件(含原程序)和JAIN-SIP库. 从SUN网站下载并安装JMF,JavaMail,JAF. 2.用NetBeans5.0编译和运行(Windows环境) 编辑build.windows.properties,设置其中java.home,jmf.home,tomcat.home,mail.home,jaf.home到合适值. 编辑build.xml,设置第二行为<property file="build.windows.properties"/>. 启动NetBeans5.0,使用AntScript形式打开本项目,运行build中的all项目,再运行build中的messenger项目,可看见<NIST Message>画面出现. 3.和NIST-SIP Proxy连接启动<NIST-SIP Proxy>. 打开<NIST Message>的Configuration画面,在URI中输入<NIST-SIP Proxy>中存在的用户ID(格式:XXX@XXX2.XXX3).在ProxyAddress和ProxyPort中输入<NIST-SIP Proxy>的地址信息.执行Register菜单命令.执行AddContact按钮命令,输入<NIST-SIP Proxy>中存在的另一用户ID(格式:XXX@XXX2.XXX3). 4.实时音频呼叫时的SIP消息分析准备: 启动一个<NIST-SIP Proxy>,地址192.168.1.2:4000; 启动两个<NIST Message>,dog@192.168.1.2:3250和cat@192.168.1.3:9385;在configuration中设为Realtime选项并设置PROXY参数为192.168.1.2:4000. 在dog的通讯录中双击cat打开对话框,单击Audio按钮,cat会弹出确认框,单打ACCEPT按钮接受邀请. 一段时间后在henrik处的对话框中单击STOP按钮关闭这次通话. 以下分析<NIST-SIP Proxy>的LOG和两个<NIST Message>的LOG,<NIST Message>的LOG可从NetBeans的TraceWindowCOPY来. From To Command Message 说明 dog Proxy INVITE INVITE sip:cat@nist.gov;transport=udp SIP/2.0Call-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 2Contact: <sip:dog@192.168.1.2:3250;transport=udp>Content-Type: application/sdpContent-Length: 109 v=0o=dog 878212 879581 IN IP4 192.168.1.2s=-c=IN IP4 192.168.1.2t=0 0m=audio 8376 RTP/AVP 5 4 3 0 Content中给出了dog的音频通道信息 Proxy dog Trying SIP/2.0 100 TryingCall-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 2Content-Length: 0 Proxy cat INVITE sip:cat@192.168.1.3:9385 SIP/2.0Call-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bK83376d784f617e7d1326103a2cf0a9b4,SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 1Contact: <sip:dog@192.168.1.2:3250;transport=udp>Content-Type: application/sdpRecord-Route: <sip:192.168.1.2:4000>Content-Length: 109 Proxy向cat转送上一消息 cat Proxy Ringing Response 180 Ringing received:SIP/2.0 180 RingingCall-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bKc202584ca6e6447cf37713092e2d109b,SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Content-Length: 0 此时cat处弹出接受确认框. Proxy dog SIP/2.0 180 RingingCall-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Content-Length: 0 Proxy向dog转送上一消息 cat Proxy OK SIP/2.0 200 OKCall-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>;tag=2312Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bKc202584ca6e6447cf37713092e2d109b,SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe40eac54e444c4faec30b0df09ddd55cMax-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Contact: <sip:cat@192.168.1.3:9385;transport=udp>Content-Type: application/sdpContent-Length: 99 v=0o=cat 7787 9156 IN IP4 192.168.1.3s=-c=IN IP4 192.168.1.3t=0 0m=audio 2262 RTP/AVP 5 cat的接受确认框被确认(按OK按钮). 注意Content的内容和dog的Invite消息已经不同,已经是cat的音频通道信息了. Proxy向dog转送上一消息 dog Proxy ACK ACK sip:192.168.1.2:4000;transport=udp SIP/2.0Call-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 ACKFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>;tag=2312Via: SIP/2.0/UDP 192.168.1.2:3250;branch=-2015693566Max-Forwards: 2Route: <sip:cat@192.168.1.3:9385;transport=udp>Content-Length: 0 Proxy向cat转送上一消息 dog侧 Get chat session: cat@nist.govcallee cat@nist.gov:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,229,163,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with cat@nist.gov,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,30,742x396,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status Ringing...Local listening audio port : 8376Remote listening audio port : 2262Init UDP TransmitterThreshold enabled : truebuf length : 0minimum Threshold : 2147483647 - Waiting for RTP data to arrive... - Open RTP session for: Address: 192.168.1.3 localPort: 8376 destPort : 2262 Time To Live: 1- number of capture devices: 2Track 0 is set to transmit as: dvi/rtp, 22050.0 Hz, 4-bit, MonoCreated RTP session: 192.168.1.3 dest 2262Media Transmission started!!!Get chat session: cat@nist.govcallee cat@nist.gov:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,229,163,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with cat@nist.gov,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,30,742x396,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call - Received new RTP stream: dvi/rtp, 22050.0 Hz, 4-bit, Mono The sender of this stream had yet to be identified.1format list : 1mpegControl::::null - A new participant had just joined: zhou@hp9010mpegControl::::nullmpegControl::::nullmpegControl::::nullmpegControl::::nullClass for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@6db33cClass for componentnull - A new participant had just joined: zhou@hp9010 - The previously unidentified stream dvi/rtp, 22050.0 Hz, 4-bit, Mono had now been identified as sent by: zhou@hp9010 dog和cat进行直接RTP通信,不经过Proxy cat侧 - Waiting for RTP data to arrive...from="192.168.1.2:4000" to="192.168.1.3:9385" time="1141344698968" isSender="false" transactionId="z9hg4bk97119ae84b4364caad83f7759147d4a5" callId="3a82d615f5028ce47d53e40c363e48c9@192.168.1.2" firstLine="ACK sip:cat@192.168.1.3:9385 SIP/2.0" - Received new RTP stream: dvi/rtp, 22050.0 Hz, 4-bit, Mono The sender of this stream had yet to be identified.debugLine="0" ><![CDATA[ACK sip:cat@192.168.1.3:9385 SIP/2.0Call-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 1 ACKFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>;tag=2312Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bK97119ae84b4364caad83f7759147d4a5,SIP/2.0/UDP 192.168.1.2:3250;branch=-2015693566Max-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Content-Length: 0 ]]></message> Init UDP TransmitterThreshold enabled : truebuf length : 0minimum Threshold : 2147483647 - Open RTP session for: Address: 192.168.1.2 localPort: 2262 destPort : 8376 Time To Live: 1- number of capture devices: 21format list : 1mpegControl::::nullTrack 0 is set to transmit as: dvi/rtp, 22050.0 Hz, 4-bit, MonoCreated RTP session: 192.168.1.2 dest 8376mpegControl::::nullmpegControl::::nullmpegControl::::nullmpegControl::::nullClass for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@22ab57Class for componentnullMedia Transmission started!!!Get chat session: dog@nist.govcallee dog@nist.gov:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with dog@nist.gov,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,30,742x396,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call - A new participant had just joined: zhou@desktop - A new participant had just joined: zhou@desktop - The previously unidentified stream dvi/rtp, 22050.0 Hz, 4-bit, Mono had now been identified as sent by: zhou@desktop dog和cat进行直接RTP通信,不经过Proxy. dog Proxy BYE BYE sip:192.168.1.2:4000;transport=udp SIP/2.0Call-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 2 BYEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>;tag=2312Via: SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe553a3a2689bc91df1ec554e21f51309Max-Forwards: 2Route: <sip:cat@192.168.1.3:9385;transport=udp>Content-Length: 0 dog册按下STOP按钮.消息发出后RTP读写终止. Proxy向cat转送上一消息 cat Proxy OK SIP/2.0 200 OKCall-ID: 3a82d615f5028ce47d53e40c363e48c9@192.168.1.2CSeq: 2 BYEFrom: <sip:dog@nist.gov>;tag=9242To: <sip:cat@nist.gov>;tag=2312Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bK26c72e5e7e444f3482fcce6ae0cc9451,SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKe553a3a2689bc91df1ec554e21f51309Max-Forwards: 2Content-Length: 0 消息发出后RTP读写终止. Proxy向dog转送上一消息如果两边的<NIST Message>的Configuration设置中都使用了MessageAudio(非RealtimeAudio),那么上面表格中的RTP通信部分要给下面样子的带有音频数据的SIP消息替,这种SIP消息还是要走SIPProxy的(因为是用SIP消息来传送音频数据,所以即使在同一台机器上开两个<NIST Message>互相通信,仍然没有问题.). ************************************************************************************************ Request MESSAGE received:MESSAGE sip:192.168.1.2:4000;transport=udp SIP/2.0Call-ID: 34fd5cabb7b241ab16a8dfd3ea72bd30@192.168.1.26CSeq: 4 MESSAGEFrom: <sip:dog@nist.gov>;tag=832To: <sip:cat@nist.gov>;tag=8664Via: SIP/2.0/UDP 192.168.1.2:3250;branch=z9hG4bKc5f13997a614055784d833a491b18038Max-Forwards: 2Route: <sip:cat@192.168.1.3:9385;transport=udp>Content-Type: audio/x-gsmContact: <sip:dog@192.168.1.2:3250;transport=udp>Content-Length: 6435 ?jT?P ??m??@6?m???@6?m??? 6?m??И?%??@6?m?? 6?m??d 6?m??? 6?m?????Y?@6?m?? 6?m??@6?m??? 6?m???raZ?`6?m??@6?m??? 6?m??@6?m???Wr]?r 6?m???@6?m??@6?m?? 6?m???z?Z?`6?m?? 6?m??|`6?m??? 6?m???r\?@6?m...................................... ************************************************************************************************ 如果cat拒绝了dog的邀请,则如下: From To Command Message 说明 dog Proxy INVITE INVITE sip:cat@nist.gov;transport=udp SIP/2.0Call-ID: 1cf141e7e76a90905602764164de10da@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=8040To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:9266;branch=z9hG4bK47cce90fb03443d2767efcf77d4e2b80Max-Forwards: 2Contact: <sip:dog@192.168.1.2:9266;transport=udp>Content-Type: application/sdpContent-Length: 109 v=0o=dog 940810 942179 IN IP4 192.168.1.2s=-c=IN IP4 192.168.1.2t=0 0m=audio 6836 RTP/AVP 5 4 3 0 Content中给出了dog的音频通道信息 Proxy dog Trying SIP/2.0 100 TryingCall-ID: 1cf141e7e76a90905602764164de10da@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=8040To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:9266;branch=z9hG4bK47cce90fb03443d2767efcf77d4e2b80Max-Forwards: 2Content-Length: 0 Proxy cat Proxy向cat转送上一消息 cat Proxy Ringing SIP/2.0 180 RingingCall-ID: 1cf141e7e76a90905602764164de10da@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=8040To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bK0437f73dd530f56022856f57cc55c49e,SIP/2.0/UDP 192.168.1.2:9266;branch=z9hG4bK47cce90fb03443d2767efcf77d4e2b80Max-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Content-Length: 0 此时cat处弹出接受确认框. Proxy dog Proxy向dog转送上一消息 cat Proxy Busy SIP/2.0 486 Busy hereCall-ID: 1cf141e7e76a90905602764164de10da@192.168.1.2CSeq: 1 INVITEFrom: <sip:dog@nist.gov>;tag=8040To: <sip:cat@nist.gov>Via: SIP/2.0/UDP 192.168.1.2:4000;branch=z9hG4bK0437f73dd530f56022856f57cc55c49e,SIP/2.0/UDP 192.168.1.2:9266;branch=z9hG4bK47cce90fb03443d2767efcf77d4e2b80Max-Forwards: 1Record-Route: <sip:192.168.1.2:4000>Call-Info: <http://www.google.com>Content-Length: 0 cat的确认框被拒绝(按Cancel按钮). Proxy向dog转送上一消息

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huashan1122@163.com(游客)发表评论于2008/12/17 2:09:31

请教您个问题，我再重新解压jain-sip-1.2以后，例子程序都运行不了了，说是 Target 'Shootme' does not exist in the project "null" 以前没出现过这个问题，不知道怎么解决

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飞翔的云(游客)发表评论于2008/10/10 11:38:44

为什么当sip采用udp传输时，其不会记udp包的长度，当时如何采用tcp传输时，会记住tcp包的长度谢谢

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yxskkk(游客)发表评论于2008/8/5 12:13:26

以下引用不留名(游客)在2007-8-1 14:08:12的评论:javax.sip.SipException: Connection refused: connect at gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:866) at gov.nist.applet.phone.ua.MessengerManager.deRegister(MessengerManager.java:565) at gov.nist.applet.phone.ua.MessengerManager.unRegisterAndReRegister(MessengerManager.java:523) at gov.nist.applet.phone.ua.gui.NISTMessengerApplet.start(NISTMessengerApplet.java:241) at sun.applet.AppletPanel.run(Unknown Source) at java.lang.Thread.run(Unknown Source)我出的就是这个错误,不知道怎么解决了..

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yxskkk(游客)发表评论于2008/8/5 8:42:40

楼主好,我的applet中,在ClientTransaction.sendRequest()的时候一直在报SipException,Connection refused: connect因为没有相关的API,实在是找不出来了,但如果以application运行的话拨号都很正常... 以下为blog主人的回复： APPLET好象是有安全限制的,即APPLET只能和下载该APPLET的主机通讯(SOCEKT). 时间过得真快,一晃都两年半了(该贴的发布日开始).

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不留名(游客)发表评论于2007/8/1 14:08:12

javax.sip.SipException: Connection refused: connect at gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:866) at gov.nist.applet.phone.ua.MessengerManager.deRegister(MessengerManager.java:565) at gov.nist.applet.phone.ua.MessengerManager.unRegisterAndReRegister(MessengerManager.java:523) at gov.nist.applet.phone.ua.gui.NISTMessengerApplet.start(NISTMessengerApplet.java:241) at sun.applet.AppletPanel.run(Unknown Source) at java.lang.Thread.run(Unknown Source)

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不留名(游客)发表评论于2007/8/1 14:07:22

具体的是： REGISTER sip:172.20.23.100:4000 SIP/2.0Call-ID: f92f36d57b99947401cb9fdf88b8cae1@172.20.22.29CSeq: 1 REGISTERFrom: <sip:mranga@nist.gov>;tag=8544To: <sip:mranga@nist.gov>Via: SIP/2.0/TCP 172.20.22.29:3590;branch=z9hG4bKd5ddcafab069ece0474400299d1e00e3Max-Forwards: 2Expires: 0Contact: *Content-Length: 0 Could not send out the un-register request! javax.sip.SipException: Connection refused: connect

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不留名(游客)发表评论于2007/8/1 14:06:08

刚看了你的blog ，我也下载了试试，，结果消息都不能发送，如果发一个 a 字符，窗口返回消息如下： mranga@nist.gov> aYour buddy is currently unavailable 同时，我在另一台机器上访问applet页面的时候，Applet控制台报错 Could not send out the un-register request! javax.sip.SipException: Connection refused: connect

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xiaoyu(游客)发表评论于2007/3/28 20:32:42

我现在也在学习JAIN-SIP Applet Phone。请问楼主，你使用时效果怎么样？例如real time的通话质量。当我使用jain-sip-presence-proxy作为server时，我的messenger只能register,其它事的都不能做。我把proxy换成Brekeke sip server之后, 我能使用voice messaging的功能，但是还是不能使用real time和发送文本消息。不知道时怎么了？当我用applet时，连register都不能了。不知道楼主有没有碰到这种情况？

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xiaoyu(游客)发表评论于2007/3/28 20:31:28

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小雨(游客)发表评论于2007/3/28 19:52:28

请问楼主，当你运行messenger时，如果选择real time能不能通话。我只有在选择voice messaging才能听到声音。而起不管选择哪个选项都不能发送短消息

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