人在旅途--mjSip学习笔记

人在旅途发表于 2006/3/13 15:38:48

1.URL:http://www.mjsip.org/ 2.它也是用JAVA编写的,唯一要求的外部库是客户端使用的JMF(只要安装了JMF就可以了,编译和执行都不用特殊处理).它比SUN的JAIN的STACK简单,而且新,上次RELEASE估计在2005年十月份,所对应的SIP功能也比JAIN多,包括支持REFER等消息格式.它也提供源程序,源程序结构比JAIN简单多了,三个部分:包括SIP的STACK和SIP的SERVER和SIP的客户端.它也有BIN的下载,服务器和客户端配置都很简单,基本上看着配置文件内部说明就可以了.从我看到该网站到学习下栽编译测试和分析LOG基本上不用一天工作时间就完成了. 3.不过它的MAKE文件是GNU的不是ANT的,对NETBEANS来说使用不便,不过由于它结构简单,我很方便的用NETBEANS5.0生成了三个PRJ(STACK,SERVER,UA),再从源程序中拷贝JAVA文件和配置文件到这三个PRJ中,很方便地就再编译和执行成功了. 这三个NETBEANS5.0的PRJ源程序和项目文件在这里:500)this.width=500'>MjSIP.rar(只用于学习,版权问题盖不负责,请去http://www.mjsip.org/查询). 4.它的SERVER缺省运行在STATELESS中,当然也可是STATEFUL的. 5.和JAIN的SAMPLE一样,大概是JMF的缘故它的UA在通话中也有很大的延迟,大约一秒多吧. 6.可惜没有商业开发的免费许可(要购买). 7.测试了下,一次INITE对话大概要用0.2秒多点时间,其他简单的TRANSACTION只要用0.1妙左右. 8.2006年4月13日,发现各致命的错误: 当INVITE或REGISTER等消息因为认证(407,401)而被打回后,再加入了认证HEADER 再送时,消息的BRANCH值仍然和第一次的一样.这样一来就和第一次同一个TRANSACTION了(尽管程序是NEW了个新TRANSACTION,但它是用上次消息经过加工后NEW的,所以BRANCH值仍然是上次旧的),而RFC要求是两个不同的TRANSACTION,这样一来服务器就可能因为新消息仍然属于旧TRANSACTION而拒绝响应. 我在ExtendedInviteDialog.onTransFailureResponse中把旧消息的BRANCH值更新了下(在BaseMessage.refreshBranch()中增加了更新函数),测试了下INVITE或REGISTER的情况,正常.

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feige(游客)发表评论于2006/3/29 18:49:08

我根据您说的修改了配置文件，但如果用 audio_codec=PCMU 就还是没有声音，但如果是 audio_codec=GSM 打电话的一方的耳机中有吱吱的声音，但对方那边还是没有声音，我再看看，非常感谢您的帮助! 我的主要配置信息如下: #via_addr=192.168.68.13#host_port=4000transport_protocols=udp tcpoutbound_proxy=192.168.68.16:5060 from_url="2224" <sip:2224@192.168.68.16>username=2224realm=192.168.68.16passwd=debug_level=8log_path=logmax_logsize=2000 contacts_file=config/contacts.lst keepalive_time=8000 audio=yesaudio_port=4002#audio_avp=0#audio_codec=PCMUaudio_codec=GSMaudio_sample_rate=8000 audio_sample_size=1audio_frame_size=500 use_jmf=yes 没列出来的都注释掉啦以下为blog主人的回复：建议除audio=yes以外的audio项全都comment掉,让程序自己去选择SDP参数.你的问题也许是两个UA之间不能建立相同SDP参数选择,可以用两台一个牌子的PC试一下(我就是这样的,问题就不大).

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fchenscut(游客)发表评论于2006/3/29 17:15:36

1.您是在实验室里面使用保留地址192.168.*.13与192.168.*.16 进行通讯,是吗? 2.在局域网内是不用使用诸如注册(REGISTRATION)和重定向(REDIRECTION)的,你的配置文件里面的一行: #do_register=yes,前面的“＃”符号已经删除了，是吗? 我昨天是在同一台电脑上测试，通过了。你也先在同一台电脑上测试吧! 今天，我连接了同一局域网上面的两台电脑，测试通过了，接收方听到的音乐像是从其本机播放出来的一样清晰!!! 我的配置是这样的: # MjSip UA configuration file# ___________________________________________# #via_addr=127.0.0.8host_port=5080transport_protocols=udp tcp#outbound_proxy=211.66.17.147:5060 from_url="Bob" <sip:bob@211.66.17.147>username=bobrealm=211.66.17.147#passwd=pippo debug_level=8log_path=logmax_logsize=2000 contacts_file=config/contacts.lst #do_register=yes#do_unregister=yes#do_unregister_all=yes keepalive_time=8000 #call_to=sip:127.0.0.9:5090 #accept_time=0#hangup_time=20#redirect_to=sip:127.0.0.9:5090#no_offer=yes#transfer_to=sip:127.0.0.9:5090#transfer_time=10#re_invite_time=5 #recv_only=yes#send_only=yes#send_tone=yes#send_file=yes audio=yesaudio_port=4002audio_avp=0audio_codec=PCMUaudio_sample_rate=8000 audio_sample_size=1audio_frame_size=500 #video=yes#video_port=4002#video_avp=101 use_jmf=yes#use_rat=yes#use_vic=yes#bin_rat="c:\program files\mbone\rat"#bin_vic="c:\program files\mbone\vic"#bin_rat=rat#bin_vic=vic 以下引用feige(游客)在2006-3-29 10:33:17的评论:首先非常感谢您的回复!可我按照您说的做了，问题依旧。每次UA(A)打电话给UA(B)后，B一旦接起电话A端就输出到:Graphical MJSIP UA 1.0UA: REGISTRATIONUA: Registration success: 200 OKUA: RINGINGUA: ACCEPTED/CALLSelected format: LINEAR, 8000.0 Hz, 8-bit, Mono ……………………然后就陷入死循环。我看过他的源代码，就是执行到:while(player.getState()!=player.Realized);这句（我看了一下player.getState() = 200 也就是Realizing而Realized=300),但不知道为什么?过很久后A端输出:UA: Registration failure: Timeout……………………AckTransactionClient#3: changed transaction state: T_TerminatedUA: onCallAccepted()UA: ACCEPTED/CALLJMFAudioLauncher: launching JMF-Audio...

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feige(游客)发表评论于2006/3/29 10:33:17

首先非常感谢您的回复! 可我按照您说的做了，问题依旧。每次UA(A)打电话给UA(B)后，B一旦接起电话 A端就输出到: Graphical MJSIP UA 1.0 UA: REGISTRATION UA: Registration success: 200 OK UA: RINGING UA: ACCEPTED/CALL Selected format: LINEAR, 8000.0 Hz, 8-bit, Mono List of devices: 1 device 0:DirectSoundCapture MediaLocator: dsound:// Processor created Processor configured ContentDescriptor=raw.rtp Processor realized Receiver URL= rtp://:21000/audio JMFAudioLauncher: launching JMF-Audio... streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@13e75a5 : 1 sink: setOutputLocator rtp://192.168.68.16:16108/audio/1 DataSink created Dest= rtp://192.168.68.16:16108/audio/1 , RTP DataSink opened Start sending Trying to realize the player 然后就陷入死循环。我看过他的源代码，就是执行到:while(player.getState()!=player.Realized); 这句（我看了一下player.getState() = 200 也就是Realizing而Realized=300),但不知道为什么? 过很久后A端输出: UA: Registration failure: Timeout events.log文件的内容如下: SipProvider-5060: Date: 10:29:55.234 Wed 29 Mar 2006 SipProvider-5060: SipStack: mjsip stack 1.6 SipProvider-5060: new SipProvider(): 5060/udp SipProvider-5060: udp is up ExtendedInviteDialog#0: changed dialog state: D_INIT ExtendedInviteDialog#0: changed dialog state: D_WAITING TransactionServer#0: id: INVITE TransactionServer#0: created TransactionServer#0: start TransactionServer#0: changed transaction state: T_Waiting SipProvider-5060: adding SipProviderListener: INVITE SipProvider-5060: 1 listeners: INVITE, UA: REGISTRATION RegisterAgent: Registering contact <sip:2223@192.168.68.13> (it expires in 3600 secs) TransactionClient#1: id: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 TransactionClient#1: created TransactionClient#1: start TransactionClient#1: changed transaction state: T_Trying SipProvider-5060: adding SipProviderListener: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 SipProvider-5060: 2 listeners: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453, INVITE, SipProvider-5060: using transport udp SipProvider-5060: Resolving host address '192.168.68.16' SipProvider-5060: Sending message to udp:192.168.68.16:5060 10:29:56.250 Wed 29 Mar 2006, 192.168.68.16:5060/udp (368 bytes): REGISTER sip:192.168.68.16 SIP/2.0, sent 10:29:56.250 Wed 29 Mar 2006, 192.168.68.16:5060/udp (425 bytes): SIP/2.0 100 Trying, received SipProvider-5060: received new SIP message SipProvider-5060: DEBUG: transaction-id: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 SipProvider-5060: message passed to transaction: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 TransactionClient#1: changed transaction state: T_Proceeding 10:29:56.281 Wed 29 Mar 2006, 192.168.68.16:5060/udp (501 bytes): SIP/2.0 200 OK, received SipProvider-5060: received new SIP message SipProvider-5060: DEBUG: transaction-id: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 SipProvider-5060: message passed to transaction: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 TransactionClient#1: changed transaction state: T_Completed RegisterAgent: Registration success: 200 OK UA: Registration success: 200 OK ExtendedInviteDialog#0: inside refuse() ExtendedInviteDialog#0: inside refuse(403,Forbidden) ExtendedInviteDialog#0: inside respond(403,Forbidden) ExtendedInviteDialog#0: WARNING: Dialog isn't in "invited" state: cannot respond (403/D_WAITING/null) ExtendedInviteDialog#0: inside cancel() SipProvider-5060: removing SipProviderListener: INVITE SipProvider-5060: 1 listeners: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453, TransactionServer#0: changed transaction state: T_Terminated ExtendedInviteDialog#0: inside bye() ExtendedCall: calling "2224" <sip:2224@192.168.68.16> ExtendedInviteDialog#1: changed dialog state: D_INIT ExtendedInviteDialog#1: inside invite(callee,caller,contact,sdp) ExtendedInviteDialog#1: inside invite(invite) ExtendedInviteDialog#1: changed dialog state: D_INVITING ExtendedInviteDialog#1: new dialog id: 729822251965@192.168.68.13-z9hG4bK78067334-null SipProvider-5060: adding SipProviderListener: 729822251965@192.168.68.13-z9hG4bK78067334-null SipProvider-5060: 2 listeners: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453, 729822251965@192.168.68.13-z9hG4bK78067334-null, TransactionClient#2: id: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 TransactionClient#2: created TransactionClient#2: start TransactionClient#2: changed transaction state: T_Trying SipProvider-5060: adding SipProviderListener: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 SipProvider-5060: 3 listeners: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453, 729822251965@192.168.68.13-z9hG4bK78067334-null, 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758, SipProvider-5060: using transport udp SipProvider-5060: Resolving host address '192.168.68.16' SipProvider-5060: Sending message to udp:192.168.68.16:5060 10:29:59.765 Wed 29 Mar 2006, 192.168.68.16:5060/udp (563 bytes): INVITE sip:2224@192.168.68.16 SIP/2.0, sent 10:29:59.765 Wed 29 Mar 2006, 192.168.68.16:5060/udp (423 bytes): SIP/2.0 100 Trying, received SipProvider-5060: received new SIP message SipProvider-5060: DEBUG: transaction-id: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 SipProvider-5060: message passed to transaction: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 TransactionClient#2: changed transaction state: T_Proceeding ExtendedInviteDialog#1: inside onTransProvisionalResponse(tc,mdg) 10:29:59.796 Wed 29 Mar 2006, 192.168.68.16:5060/udp (439 bytes): SIP/2.0 180 Ringing, received SipProvider-5060: received new SIP message SipProvider-5060: DEBUG: transaction-id: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 SipProvider-5060: message passed to transaction: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 ExtendedInviteDialog#1: inside onTransProvisionalResponse(tc,mdg) UA: onCallRinging() UA: RINGING TransactionClient#1: Clearing timeout expired SipProvider-5060: removing SipProviderListener: 334060406272@192.168.68.13-1-REGISTER-192.168.68.13:5060-z9hG4bK65453 SipProvider-5060: 2 listeners: 729822251965@192.168.68.13-z9hG4bK78067334-null, 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758, TransactionClient#1: changed transaction state: T_Terminated 10:30:01.968 Wed 29 Mar 2006, 192.168.68.16:5060/udp (652 bytes): SIP/2.0 200 OK, received SipProvider-5060: received new SIP message SipProvider-5060: DEBUG: transaction-id: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 SipProvider-5060: message passed to transaction: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 TransactionClient#2: changed transaction state: T_Terminated SipProvider-5060: removing SipProviderListener: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758 SipProvider-5060: 1 listeners: 729822251965@192.168.68.13-z9hG4bK78067334-null, ExtendedInviteDialog#1: inside onTransSuccessResponse(729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK70758,msg) ExtendedInviteDialog#1: inside onTransSuccessResponse(tc,msg) ExtendedInviteDialog#1: changed dialog state: D_CALL SipProvider-5060: removing SipProviderListener: 729822251965@192.168.68.13-z9hG4bK78067334-null SipProvider-5060: 0 listeners: ExtendedInviteDialog#1: new dialog id: 729822251965@192.168.68.13-z9hG4bK78067334-as278e6c90 SipProvider-5060: adding SipProviderListener: 729822251965@192.168.68.13-z9hG4bK78067334-as278e6c90 SipProvider-5060: 1 listeners: 729822251965@192.168.68.13-z9hG4bK78067334-as278e6c90, AckTransactionClient#3: id: 729822251965@192.168.68.13-1-INVITE-192.168.68.13:5060-z9hG4bK53471 AckTransactionClient#3: created AckTransactionClient#3: start SipProvider-5060: using transport udp SipProvider-5060: Resolving host address '192.168.68.16' SipProvider-5060: Sending message to udp:192.168.68.16:5060 10:30:01.968 Wed 29 Mar 2006, 192.168.68.16:5060/udp (378 bytes): ACK sip:2224@192.168.68.16 SIP/2.0, sent AckTransactionClient#3: changed transaction state: T_Terminated UA: onCallAccepted() UA: ACCEPTED/CALL JMFAudioLauncher: launching JMF-Audio...

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人在旅途发表评论于2006/3/29 10:11:41

各位朋友,我碰到各问题. 当我用MjSIP的UA和其他SIPSERVER测试时,在REGISTER时返回了401(认证要求,MD5),UA添加了认证信息后再送,被SERVER接受;随后,UA发送了INVITE消息,SERVER又返回了407(认证要求,MD5),我又同样得添加了认证信息后再送,但SERVER就再也没有反映了.SERVER应该没问题,因为我用HARDPHONE试过了,同样有认证要求,但都顺利通过了,我想是MjSIP的UA和它不匹配吧,有人有这个问题的经验吗? 以下为blog主人的回复：已经解决,是BRANCH值没变的原因,请看更改后的主贴.

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fchenscut(游客)发表评论于2006/3/28 19:51:14

非常感谢blog主人的提供源码和配置的热诚。 Thanks a lot！！！

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fchenscut(游客)发表评论于2006/3/28 19:48:23

不好意思，现在好了。声音混浊的缘故是我打开了不必打开的很多播放选项。现在听到的声音很清，但有回音，延迟应该小于半秒吧!我是在同一台电脑上测试的。

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fchenscut(游客)发表评论于2006/3/28 19:41:52

连接刚成功的时候，听到的声音是很不错的。但是，渐渐地，会出现很大的钻井似的噪音，越来越大以致盖过了要传输的原音。不知何故。

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fchenscut(游客)发表评论于2006/3/28 19:30:40

接下面的那条信息。注意要把配置文件＊.cfg里面的“use_jmf=yes”前面的＃去掉。

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fchenscut(游客)发表评论于2006/3/28 19:27:49

这位朋友，我找到没能通话的原因了。下面我测试的log提示： Selected format: LINEAR, 8000.0 Hz, 8-bit, Mono (-----注意这里,原因就出在这里-----)List of devices: 1device 0:DirectSoundCaptureMediaLocator: dsound://Processor createdProcessor configuredContentDescriptor=raw.rtpProcessor realizedReceiver URL= rtp://:4000/audioJMFAudioLauncher: launching JMF-Audio...streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@1b09468 : 1sink: setOutputLocator rtp://211.66.17.147:3000/audio/1DataSink createdDest= rtp://211.66.17.147:3000/audio/1 , RTPDataSink openedStart sendingTrying to realize the playerPlayer realizedStop sendingPlayer stopped 注意到红色部分文字,程序是通过系统的mono来录音,然后再发送出去的. 所以您如果要测试的话，请打开你的windows(如果是这种系统的话)的音频属性，选择录音选项，会发现一个“mono mix”，然后选择这个来录音，就会听到你要传输的音频了。我在同一台电脑上试了一下，简单地播放一个音乐文件，可以顺利听到由UA返传回来的声音，不过很混浊难听。以下引用feige(游客)在2006-3-28 17:27:46的评论:不好意思再次打扰拉，我UA联系对方，对方的电话铃可以响，但就是无法听到双方的声音。每次都输出JMFAudioLauncher: launching JMF-Audio...streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@7be8c2 : 1sink: setOutputLocator rtp://192.168.68.16:13596/audio/1DataSink createdDest= rtp://192.168.68.16:13596/audio/1 , RTPDataSink openedStart sendingTrying to realize the player然后就停在这啦，看了他的Log后也不太清楚：…………………………………………………………………………上面是他的部分log其实我的目的是想利用MJSIP提供的电话转接功能我之前用Jain-Sip-Applet-Phone已经实现了Phone to Pc 和Pc to Phone但 Refer功能做不出来，我按照SIP中对Refer的定义写了Refer，但每次执行完毕后refer-by一方已经收到了Accepted信息，但refer-to一方死活就是没有任何反应，我感觉Accepted好像是VOIPServer发送过来的，不知搂主有没有这方面的经验，还望给予帮忙，先谢谢啦

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fchenscut(游客)发表评论于2006/3/28 19:10:43

配置文件a.cfg中有这么几行: #bin_rat="c:\program files\mbone\rat"#bin_vic="c:\program files\mbone\vic"bin_rat=ratbin_vic=vic 谁知道是干什么用的?是用来配置多媒体选项的吗? 按照上面提示的路径我找不到那些文件.请问是不是要额外安装的? 以下为blog主人的回复：这大概是和JMF不同的语音系统吧,没研究过.

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