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<channel>
	<title>Star life &#187; voip</title>
	<atom:link href="http://liuchangjun.com/tag/voip/feed/" rel="self" type="application/rss+xml" />
	<link>http://liuchangjun.com</link>
	<description>无欲速 无见小利 欲速则不达 见小利则大事不成</description>
	<lastBuildDate>Wed, 13 Jul 2011 15:00:25 +0000</lastBuildDate>
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		<item>
		<title>Howto: RTPproxy Installation Guide</title>
		<link>http://liuchangjun.com/2011/04/08/howto-rtpproxy-installation-guide/</link>
		<comments>http://liuchangjun.com/2011/04/08/howto-rtpproxy-installation-guide/#comments</comments>
		<pubDate>Fri, 08 Apr 2011 08:17:00 +0000</pubDate>
		<dc:creator>star</dc:creator>
				<category><![CDATA[Tech Tools]]></category>
		<category><![CDATA[centos]]></category>
		<category><![CDATA[kamailio]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[openser]]></category>
		<category><![CDATA[rtp]]></category>
		<category><![CDATA[rtpproxy]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://liuchangjun.com/?p=248</guid>
		<description><![CDATA[<p>The Sippy RTPproxy is a high-performance software proxy server for RTP streams that can work together with SIP Express Router (SER), OpenSIPS, Kamailio, Sippy B2BUA or reSIProcate B2BUA.</p> <p>The main purpose of RTPproxy originally had been to make the communication between SIP user agents behind NAT(s) (Network Address Translator) possible.</p> <p>Homepage: http://www.rtpproxy.org/</p> <p>1. Installation Download [...]]]></description>
			<content:encoded><![CDATA[<p>The Sippy RTPproxy is a high-performance software proxy server for RTP streams that can work together with  SIP Express Router (SER),  OpenSIPS,  Kamailio, Sippy B2BUA or  reSIProcate B2BUA.</p>
<p>The main purpose of RTPproxy originally had been to make the communication between SIP user agents behind NAT(s) (Network Address Translator) possible.</p>
<p>Homepage: <a href="http://www.rtpproxy.org/">http://www.rtpproxy.org/</a></p>
<p>1. Installation<br />
Download the package<br />
<code><br />
# cd ~<br />
# wget http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz<br />
# tar zxvf rtpproxy-1.2.1.tar.gz<br />
# cd rtpproxy-1.2.1<br />
</code><br />
Make and install<br />
<code><br />
# ./configure<br />
# make<br />
# make install<br />
</code></p>
<p>2. Running with Kamailio<br />
<code><br />
# rtpproxy -l xxx.xxx.xxx.xxx -s udp:localhost 7722 -u ftp<br />
</code></p>
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		<item>
		<title>Howto: Kamailio (OpenSER) Installation Guide</title>
		<link>http://liuchangjun.com/2011/04/08/howto-kamailio-openser-installation-guide/</link>
		<comments>http://liuchangjun.com/2011/04/08/howto-kamailio-openser-installation-guide/#comments</comments>
		<pubDate>Fri, 08 Apr 2011 08:05:45 +0000</pubDate>
		<dc:creator>star</dc:creator>
				<category><![CDATA[Tech Tools]]></category>
		<category><![CDATA[centos]]></category>
		<category><![CDATA[kamailio]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[openser]]></category>
		<category><![CDATA[rtp]]></category>
		<category><![CDATA[rtpproxy]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://liuchangjun.com/?p=246</guid>
		<description><![CDATA[<p>Kamailio (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second.</p> <p>Homepage: http://www.kamailio.org</p> <p>1. Installation Download the source package # cd ~ # wget http://www.kamailio.org/pub/kamailio/latest/src/kamailio-3.1.3_src.tar.gz # tar zxvf kamailio-3.1.3_src.tar.gz # cd kamailio-3.1.3 # make FLAVOUR=kamailio cfg Modify “modules.lst”, Remove db_mysql from the variable exclude_modules. # [...]]]></description>
			<content:encoded><![CDATA[<p>Kamailio (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second.</p>
<p>Homepage: <a href="http://www.kamailio.org">http://www.kamailio.org</a></p>
<p>1. Installation<br />
Download the source package<br />
<code><br />
# cd ~<br />
# wget http://www.kamailio.org/pub/kamailio/latest/src/kamailio-3.1.3_src.tar.gz<br />
# tar zxvf kamailio-3.1.3_src.tar.gz<br />
# cd kamailio-3.1.3<br />
# make FLAVOUR=kamailio cfg<br />
</code><br />
Modify “modules.lst”, Remove db_mysql from the variable exclude_modules.<br />
<code><br />
# make all<br />
# make install<br />
</code><br />
The Kamailio was installed in “/usr/local/sbin”, the configuration file was installed in “/usr/local/etc/kaimailio”<br />
<code><br />
    kamailio - Kamailio (OpenSER) server<br />
    kamdbctl - script to create and manage the Databases<br />
    kamctl - script to manage and control Kamailio (OpenSER) server<br />
    sercmd - CLI - command line tool to interface with Kamailio (OpenSER) server<br />
</code></p>
<p>2. Configuration<br />
Config the enviroment variables<br />
<code><br />
# cat /usr/local/etc/kamailio/kamctlrc<br />
# $Id$<br />
#<br />
# The Kamailio configuration file for the control tools.<br />
#<br />
# Here you can set variables used in the kamctl and kamdbctl setup<br />
# scripts. Per default all variables here are commented out, the control tools<br />
# will use their internal default values.</p>
<p>## your SIP domain<br />
# SIP_DOMAIN=kamailio.org<br />
SIP_DOMAIN=hzserver1</p>
<p>## chrooted directory<br />
# $CHROOT_DIR="/path/to/chrooted/directory"</p>
<p>## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, or DBTEXT, by default none is loaded<br />
# If you want to setup a database with kamdbctl, you must at least specify<br />
# this parameter.<br />
# DBENGINE=MYSQL<br />
DBENGINE=MYSQL</p>
<p>## database host<br />
# DBHOST=localhost<br />
DBHOST=localhost</p>
<p>## database name (for ORACLE this is TNS name)<br />
# DBNAME=openser<br />
DBNAME=openser</p>
<p># database path used by dbtext or db_berkeley<br />
# DB_PATH="/usr/local/etc/kamailio/dbtext"</p>
<p>## database read/write user<br />
# DBRWUSER=openser<br />
DBRWUSER=openser</p>
<p>## password for database read/write user<br />
# DBRWPW="openserrw"<br />
DBRWPW="openserrw"</p>
<p>## database read only user<br />
# DBROUSER=openserro<br />
DBROUSER=openserro</p>
<p>## password for database read only user<br />
# DBROPW=openserro<br />
DBROPW=openserro</p>
<p>## database super user (for ORACLE this is 'scheme-creator' user)<br />
# DBROOTUSER="root"<br />
DBROOTUSER="root"</p>
<p># user name column<br />
# USERCOL="username"<br />
USERCOL="username"</p>
<p># SQL definitions<br />
# If you change this definitions here, then you must change them<br />
# in db/schema/entities.xml too.<br />
# FIXME</p>
<p># FOREVER="2020-05-28 21:32:15"<br />
# DEFAULT_ALIASES_EXPIRES=$FOREVER<br />
# DEFAULT_Q="1.0"<br />
# DEFAULT_CALLID="Default-Call-ID"<br />
# DEFAULT_CSEQ="13"<br />
# DEFAULT_LOCATION_EXPIRES=$FOREVER</p>
<p># Program to calculate a message-digest fingerprint<br />
# MD5="md5sum"</p>
<p># awk tool<br />
# AWK="awk"</p>
<p># If you use a system with a grep and egrep that is not 100% gnu grep compatible,<br />
# e.g. solaris, install the gnu grep (ggrep) and specify this below.<br />
#<br />
# grep tool<br />
# GREP="grep"</p>
<p># egrep tool<br />
# EGREP="egrep"</p>
<p># sed tool<br />
# SED="sed"</p>
<p># tail tool<br />
# LAST_LINE="tail -n 1"</p>
<p># expr tool<br />
# EXPR="expr"</p>
<p># Describe what additional tables to install. Valid values for the variables<br />
# below are yes/no/ask. With ask (default) it will interactively ask the user<br />
# for an answer, while yes/no allow for automated, unassisted installs.<br />
#</p>
<p># If to install tables for the modules in the EXTRA_MODULES variable.<br />
# INSTALL_EXTRA_TABLES=ask</p>
<p># If to install presence related tables.<br />
# INSTALL_PRESENCE_TABLES=ask</p>
<p># Define what module tables should be installed.<br />
# If you use the postgres database and want to change the installed tables, then you<br />
# must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly in the<br />
# kamdbctl.base script.</p>
<p># Kamailio standard modules<br />
# STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo<br />
#                   alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher<br />
#                   dialplan"</p>
<p># Kamailio extra modules<br />
# EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist htable purple"</p>
<p>## type of aliases used: DB - database aliases; UL - usrloc aliases<br />
## - default: none<br />
# ALIASES_TYPE="DB"</p>
<p>## control engine: FIFO or UNIXSOCK<br />
## - default FIFO<br />
# CTLENGINE="FIFO"</p>
<p>## path to FIFO file<br />
# OSER_FIFO="FIFO"</p>
<p>## check ACL names; default on (1); off (0)<br />
# VERIFY_ACL=1</p>
<p>## ACL names - if VERIFY_ACL is set, only the ACL names from below list<br />
## are accepted<br />
# ACL_GROUPS="local ld int voicemail free-pstn"</p>
<p>## verbose - debug purposes - default '0'<br />
# VERBOSE=1<br />
VERBOSE=1</p>
<p>## do (1) or don't (0) store plaintext passwords<br />
## in the subscriber table - default '1'<br />
# STORE_PLAINTEXT_PW=0</p>
<p>## OPENSER START Options<br />
## PID file path - default is: /var/run/kamailio.pid<br />
# PID_FILE=/var/run/kamailio.pid<br />
PID_FILE=/var/run/kamailio.pid</p>
<p>## Extra start options - default is: not set<br />
# example: start Kamailio with 64MB share memory: STARTOPTIONS="-m 64"<br />
# STARTOPTIONS=<br />
STARTOPTIONS="-u ftp"<br />
</code><br />
Add the following lines in the config file.<br />
<code><br />
# cat /usr/local/etc/kamailio/kamailio.cfg<br />
......<br />
#!define WITH_MYSQL<br />
#!define WITH_AUTH<br />
#!define WITH_USRLOCDB<br />
#!define WITH_NAT<br />
......<br />
</code><br />
Create MySQL database (Do change the passwords for these two users immediately after the database is created)<br />
<code><br />
# /usr/local/sbin/kamdbctl create<br />
</code></p>
<p>3. Start the Kamailio<br />
<code><br />
# /usr/local/sbin/kamailio<br />
</code></p>
<p>4. Monitor the Kamailio<br />
<code><br />
# /usr/local/sbin/kamctl moni<br />
</code></p>
<p>5. Add user<br />
<code><br />
# kamctl add 80001 80001<br />
</code></p>
<p>Refer to:<br />
<a href="http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git">http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git</a></p>
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		</item>
		<item>
		<title>Howto: Voip Performance test &#8211; SIPp</title>
		<link>http://liuchangjun.com/2011/02/21/howto-sip-performance-test/</link>
		<comments>http://liuchangjun.com/2011/02/21/howto-sip-performance-test/#comments</comments>
		<pubDate>Mon, 21 Feb 2011 06:26:21 +0000</pubDate>
		<dc:creator>star</dc:creator>
				<category><![CDATA[Tech Tools]]></category>
		<category><![CDATA[Howto]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[test]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://liuchangjun.com/?p=241</guid>
		<description><![CDATA[<p>Homepage: http://sipp.sourceforge.net/</p> <p>测试软件 SIPp v3.2-TLS-PCAP, version unknown, built Feb 17 2011, 10:19:21.</p> <p>安装编译： 下载源代码后重新编译 # wget http://sourceforge.net/projects/sipp/files/sipp/3.2/sipp.svn.tar.gz/download?use_mirror=ncu # tar zxvf sipp.svn.tar.gz # cd sipp.svn # make pcapplay_ossl</p> <p>备注：测试RTP需要pcap包，测试认证需要OpenSSL包，e.g. * C++ Compiler * curses or ncurses library * For authentication and TLS support: OpenSSL &#62;= 0.9.8 * For pcap play support: libpcap and libnet * For [...]]]></description>
			<content:encoded><![CDATA[<p>Homepage: <a href="http://sipp.sourceforge.net/">http://sipp.sourceforge.net/</a></p>
<p>测试软件<br />
SIPp v3.2-TLS-PCAP, version unknown, built Feb 17 2011, 10:19:21.</p>
<p>安装编译：<br />
下载源代码后重新编译<br />
# wget http://sourceforge.net/projects/sipp/files/sipp/3.2/sipp.svn.tar.gz/download?use_mirror=ncu<br />
# tar zxvf sipp.svn.tar.gz<br />
# cd sipp.svn<br />
# make pcapplay_ossl</p>
<p>备注：测试RTP需要pcap包，测试认证需要OpenSSL包，e.g.<br />
* C++ Compiler<br />
* curses or ncurses library<br />
* For authentication and TLS support: OpenSSL &gt;= 0.9.8<br />
* For pcap play support: libpcap and libnet<br />
* For distributed pauses: Gnu Scientific Libraries</p>
<p>测试脚本命令<br />
sipp -sf meeting.xml -inf meeting.csv -l 60 -m 60 -i 192.168.1.67 192.168.1.40:5060</p>
<p>测试流程<br />
1. 创建会议接入号：10000；<br />
2. 创建会议：123456，密码：123；</p>
<p>SIPp UAC               MCU<br />
|(1) INVITE         |<br />
|&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&gt;|<br />
|(2) 100 (optional) |<br />
|&lt;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;|<br />
|(3) 183 (optional) |<br />
|&lt;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;|<br />
|(4) 200            |<br />
|&lt;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;|<br />
|(5) ACK            |<br />
|&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&gt;|<br />
|                   |<br />
|(6) RFC2833 DIGIT  |<br />
|    (123456#)      |<br />
|                   |<br />
|==================&gt;|<br />
|(7) RFC2833 DIGIT  |<br />
|                   |<br />
|      (123#)       |<br />
|==================&gt;|<br />
|                   |<br />
|(8) RTP send (600s)|<br />
|==================&gt;|<br />
|                   |<br />
|(9) BYE            |<br />
|&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&gt;|<br />
|(10) 200           |<br />
|&lt;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;|</p>
<p>相关脚本：<br />
gennum.sh<br />
<code><br />
#!/bin/bash<br />
echo "SEQUENTIAL" &gt; meeting.csv<br />
i=80000<br />
while [ $i != 89999  ]<br />
do<br />
i=$(($i+1))<br />
echo "$i;10000" &gt;&gt;meeting.csv<br />
#  j=$(($i+1))<br />
#echo "$i;[authentication username=$i password=$i]" &gt;&gt;meeting.csv<br />
done<br />
</code></p>
<p>meeting.csv<br />
<code><br />
SEQUENTIAL<br />
80001;10000<br />
80002;10000<br />
80003;10000<br />
</code></p>
<p>meeting.sh<br />
<code><br />
./sipp -sf meeting.xml -inf meeting.csv -p 10000 -l 60 -m 10000 -i 192.168.1.67 192.168.1.40:15060<br />
</code></p>
<p>meeting.xml<br />
<a href="http://sipp.sourceforge.net/doc/uac_pcap.xml">参考uac_pcap.xml</a></p>
<p>Refer to:<br />
<a href="http://blogold.chinaunix.net/u/4631/showart_220923.html">SIP压力测试最好的工具，SIPp的安装与使用</a></p>
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		</item>
		<item>
		<title>X-Lite &#8211; free SIP phone (VoIP) client under Linux</title>
		<link>http://liuchangjun.com/2010/01/13/x-lite-start/</link>
		<comments>http://liuchangjun.com/2010/01/13/x-lite-start/#comments</comments>
		<pubDate>Wed, 13 Jan 2010 08:09:26 +0000</pubDate>
		<dc:creator>star</dc:creator>
				<category><![CDATA[Tech Tools]]></category>
		<category><![CDATA[deepin]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[Ubuntu]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://liuchangjun.com/?p=182</guid>
		<description><![CDATA[<p>成功安装使用 X-Lite 2.0 for Linux，效果不错的，就是配置要注意些。 该版本只有语音而没有支持图像，看来其开发的重点还是 Windows/Mac。</p> <p>备注: Linux Deepin 9.12 (Ubuntu 2009-11-03) 缺少了 libstdc++.so.5 库，按照下面的方法安装: Getting 32-bit libstdc++.so.5 in Karmic Koala on a 64-bit system 看不到的同学直接下载安装: http://packages.ubuntu.com/jaunty/i386/libstdc++5/download</p> <p>&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8211; X-Lite 是一种特有VoIP软件电话的免费软件，使用会话发起协议。X-Lite 由位于温哥华的一家CounterPath软件公司所开发。 X-Lite 目前主要有两种不同的产品。X-Lite 2.0 使用于Linux平台，是使用旧式的 X-Pro code base, 而 X-Lite 3.0 是使用于 Windows 和 Mac OS X，使用 eyeBeam code base. X-Lite 2.0 只有声音，没有影像。X-Lite 3.0 [...]]]></description>
			<content:encoded><![CDATA[<p>成功安装使用 X-Lite 2.0 for Linux，效果不错的，就是配置要注意些。<br />
该版本只有语音而没有支持图像，看来其开发的重点还是 Windows/Mac。</p>
<p>备注:<br />
Linux Deepin 9.12 (Ubuntu 2009-11-03) 缺少了 libstdc++.so.5 库，按照下面的方法安装: <a href="http://agentzlerich.blogspot.com/2009/11/getting-32-bit-libstdcso5-in-karmic.html">Getting 32-bit libstdc++.so.5 in Karmic Koala on a 64-bit system</a><br />
看不到的同学直接下载安装: <a href=" http://packages.ubuntu.com/jaunty/i386/libstdc++5/download">http://packages.ubuntu.com/jaunty/i386/libstdc++5/download</a></p>
<p>&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8211;<br />
X-Lite 是一种特有VoIP软件电话的免费软件，使用会话发起协议。X-Lite 由位于温哥华的一家CounterPath软件公司所开发。<br />
X-Lite 目前主要有两种不同的产品。X-Lite 2.0 使用于Linux平台，是使用旧式的 X-Pro code base, 而 X-Lite 3.0 是使用于 Windows 和 Mac OS X，使用 eyeBeam code base. X-Lite 2.0 只有声音，没有影像。X-Lite 3.0 则兼具声音和影像。<br />
2005年，本产品荣获 Internet Telephony 杂志的年度最佳产品.<br />
<img src="http://www.counterpath.com/assets/images/191/x-lite_banner_20091207.jpg" alt="X-Lite" /></p>
<p>Refer to:<br />
<a href="http://zh.wikipedia.org/wiki/X-Lite">X-Lite</a><br />
<a href="http://zh.wikipedia.org/zh-cn/VoIP%E8%BB%9F%E9%AB%94%E7%9A%84%E6%AF%94%E8%BC%83">VoIP软件的比较</a></p>
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		</item>
		<item>
		<title>Google Voice 注册成功，号码比较好记，正常呼出和呼入</title>
		<link>http://liuchangjun.com/2009/07/23/google-voice-%e6%b3%a8%e5%86%8c%e6%88%90%e5%8a%9f%ef%bc%8c%e5%8f%b7%e7%a0%81%e6%af%94%e8%be%83%e5%a5%bd%e8%ae%b0/</link>
		<comments>http://liuchangjun.com/2009/07/23/google-voice-%e6%b3%a8%e5%86%8c%e6%88%90%e5%8a%9f%ef%bc%8c%e5%8f%b7%e7%a0%81%e6%af%94%e8%be%83%e5%a5%bd%e8%ae%b0/#comments</comments>
		<pubDate>Thu, 23 Jul 2009 03:23:42 +0000</pubDate>
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		<guid isPermaLink="false">http://liuchangjun.com/?p=133</guid>
		<description><![CDATA[<p>以下内容仅供测试研究。</p> <p>Step 1:</p> <p>Apply one Google Voice account, e.g. one free US phone number. You should use the proxy to access Google Voice, it will spend several days.</p> <p>For the China subscriber, refer to: http://google.org.cn/2009/07/15/how-to-signup-google-voice-in-china-mainland/</p> <p>Step 2:</p> <p>Apply one Gizmo5 account, e.g. one SIP Phone number, one user name and password. It will be taken [...]]]></description>
			<content:encoded><![CDATA[<p>以下内容仅供测试研究。</p>
<p>Step 1:</p>
<p>Apply one Google Voice account, e.g. one free US phone number. You should use the proxy to access Google Voice, it will spend several days.</p>
<p>For the China subscriber, refer to: http://google.org.cn/2009/07/15/how-to-signup-google-voice-in-china-mainland/</p>
<p>Step 2:</p>
<p>Apply one Gizmo5  account, e.g. one SIP Phone number, one user name and password. It will be taken affect immediately, but a slug of network is from China.</p>
<p>Refer to: http://www.jrin.net/2009_07_26/use-gizmo5-for-free-calls-with-google-voice</p>
<p>Step 3:</p>
<p>Make a connection between Google Voice and Gizmo5 accounts.</p>
<p>Refer to: http://www.jrin.net/2009_07_26/use-gizmo5-for-free-calls-with-google-voice</p>
<p><span id="more-133"></span></p>
<p>Created 2009/07/23</p>
<p>关注google voice一段时间了，昨天终于注册成功，号码是(518) 594-1818。<br />
参考下面的文章：<a href="http://google.org.cn/2009/07/15/how-to-signup-google-voice-in-china-mainland/">国内注册 Google Voice 的方法</a></p>
<p>考虑到www.virtualphoneline.com的号码只有24天试用期，从ipkall.com重新注册了个号码，免费而且无期限:-)<br />
各位TX谁要是也申请了，可以给偶语音留言。持续关注&#8230;&#8230;</p>
<p>Updated 2009/07/28<br />
从<a href="http://google.org.cn">谷奥</a>那里获知通过Gizmo5和Google Voice一起使用可以完整的打入和打出电话，根据网上的资料操作成功，呼叫时间未测试。<br />
参考下面的文章：<br />
最初的消息来源：<a href="http://google.org.cn/posts/gizmovoice.html">Gizmo5+Google Voice=GizmoVoice</a><br />
实施的详细资料：<a href="http://www.jrin.net/2009_07_26/use-gizmo5-for-free-calls-with-google-voice">Use Gizmo5 for free calls with Google Voice</a></p>
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		<title>Howto: tcpdump user guide</title>
		<link>http://liuchangjun.com/2008/08/12/tcpdump-user-guide/</link>
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		<pubDate>Tue, 12 Aug 2008 02:16:56 +0000</pubDate>
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		<guid isPermaLink="false">http://star.72pines.com/2008/08/12/tcpdump-user-guide/</guid>
		<description><![CDATA[<p>示例1：指定网卡，抓包到文件 # tcpdump -i hme0 -w /tmp/sms.cap -i xxx Interface -w xxx write into file</p> <p>示例2：指定网卡，指定端口，屏幕打印，抓包完整内容 # tcpdump -A -n -s 0 port 5060 -i eth1 -A print ASCII -n keep IP address -s 0 capture all data for each packet port xxx captured the port</p> <p>1. 命令格式： tcpdump [ -adeflnNOpqStvx ] [ -c 数量 [...]]]></description>
			<content:encoded><![CDATA[<p>示例1：指定网卡，抓包到文件<br />
# tcpdump -i hme0 -w /tmp/sms.cap<br />
-i xxx   Interface<br />
-w xxx   write into file</p>
<p>示例2：指定网卡，指定端口，屏幕打印，抓包完整内容<br />
# tcpdump -A -n -s 0 port 5060 -i eth1<br />
-A   print ASCII<br />
-n   keep IP address<br />
-s 0   capture all data for each packet<br />
port xxx   captured the port</p>
<p>1. 命令格式：<br />
tcpdump [ -adeflnNOpqStvx ] [ -c 数量 ] [ -F 文件名 ]<br />
[ -i 网络接口 ] [ -r 文件名] [ -s snaplen ]<br />
[ -T 类型 ] [ -w 文件名 ] [表达式 ]</p>
<p>2. tcpdump的选项介绍<br />
-a 　　　将网络地址和广播地址转变成名字；<br />
-d 　　　将匹配信息包的代码以人们能够理解的汇编格式给出；<br />
-dd 　　　将匹配信息包的代码以c语言程序段的格式给出；<br />
-ddd 　　　将匹配信息包的代码以十进制的形式给出；<br />
-e 　　　在输出行打印出数据链路层的头部信息；<br />
-f 　　　将外部的Internet地址以数字的形式打印出来；<br />
-l 　　　使标准输出变为缓冲行形式；<br />
-n 　　　不把网络地址转换成名字；<br />
-t 　　　在输出的每一行不打印时间戳；<br />
-v 　　　输出一个稍微详细的信息，例如在ip包中可以包括ttl和服务类型的信息；<br />
-vv 　　　输出详细的报文信息；<br />
-c 　　　在收到指定的包的数目后，tcpdump就会停止；<br />
-F 　　　从指定的文件中读取表达式,忽略其它的表达式；<br />
-i 　　　指定监听的网络接口；<br />
-r 　　　从指定的文件中读取包(这些包一般通过-w选项产生)；<br />
-w 　　　直接将包写入文件中，并不分析和打印出来；<br />
-T 　　　将监听到的包直接解释为指定的类型的报文，常见的类型有rpc（远程过程调用）和snmp（简单网络管理协议）</p>
<p>Refer to:<br />
<a href="http://www.chinaunix.net/jh/29/44464.html">tcpdump使用说明</a></p>
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