文件列表:

255708\sip原代码\sip\test\netcat\200\200 (278, 2000-09-16)
255708\sip原代码\sip\ua\Tone\3beeps (24282, 2001-06-22)
255708\sip原代码\sip\drivers\tone\3beeps (24282, 2001-08-31)
255708\sip原代码\sip\gua\gui\ABOUT-NLS (24053, 2002-11-26)
255708\sip原代码\sip\test\netcat\ack\ack (413, 2000-09-16)
255708\sip原代码\sip\test\netcat\bad_to\bad_to (307, 2000-09-16)
255708\sip原代码\sip\test\netcat\branch_via\branch_via (325, 2000-09-16)
255708\sip原代码\sip\test\netcat\cancel\cancel (246, 2000-09-16)
255708\sip原代码\sip\test\netcat\cancel\cancel2 (314, 2000-09-16)
255708\sip原代码\sip\ua\ChangeLog (270, 2001-05-13)
255708\sip原代码\sip\gua\gui\ChangeLog (357, 2002-11-26)
255708\sip原代码\sip\test\netcat\invite_all\crlfTmp (560, 2000-09-16)
255708\sip原代码\sip\ua\Tone\cwbeep (8094, 2000-07-21)
255708\sip原代码\sip\drivers\tone\cwbeep (8094, 2001-08-31)
255708\sip原代码\sip\ua\Tone\dialtone (116640, 2000-06-09)
255708\sip原代码\sip\drivers\tone\dialtone (116640, 2001-08-31)
255708\sip原代码\sip\test\netcat\empty_msg\empty_msg2 (56, 2000-09-16)
255708\sip原代码\sip\Doc\CVS\Entries (181, 2003-04-03)
255708\sip原代码\sip\b2b\controlState\CVS\Entries (838, 2003-04-03)
255708\sip原代码\sip\base\doc++\CVS\Entries (242, 2003-04-03)
255708\sip原代码\sip\base\together\CVS\Entries (395, 2003-04-03)
255708\sip原代码\sip\drivers\tone\CVS\Entries (218, 2003-04-03)
255708\sip原代码\sip\examples\CVS\Entries (565, 2003-04-03)
255708\sip原代码\sip\gua\controlState\CVS\Entries (1098, 2003-04-03)
255708\sip原代码\sip\gua\gui\src\pixmaps\CVS\Entries (126, 2003-04-03)
255708\sip原代码\sip\libsipua\uaState\CVS\Entries (1437, 2003-04-03)
255708\sip原代码\sip\sipstack\CVS\Entries (18162, 2003-04-03)
255708\sip原代码\sip\test\messages\other\CVS\Entries (173, 2003-04-03)
255708\sip原代码\sip\test\messages\torture\CVS\Entries (2497, 2003-04-03)
255708\sip原代码\sip\test\netcat\200\CVS\Entries (52, 2003-04-03)
255708\sip原代码\sip\test\netcat\ack\CVS\Entries (52, 2003-04-03)
255708\sip原代码\sip\test\netcat\bad_to\CVS\Entries (55, 2003-04-03)
255708\sip原代码\sip\test\netcat\branch_via\CVS\Entries (59, 2003-04-03)
255708\sip原代码\sip\test\netcat\cancel\CVS\Entries (109, 2003-04-03)
255708\sip原代码\sip\test\netcat\empty_msg\CVS\Entries (115, 2003-04-03)
255708\sip原代码\sip\test\netcat\miss_to\CVS\Entries (56, 2003-04-03)
255708\sip原代码\sip\test\netcat\mess_via\CVS\Entries (57, 2003-04-03)
255708\sip原代码\sip\test\netcat\min_invite\CVS\Entries (59, 2003-04-03)
255708\sip原代码\sip\test\netcat\miss_invite\CVS\Entries (60, 2003-04-03)
255708\sip原代码\sip\test\netcat\incorrect_from\CVS\Entries (63, 2003-04-03)
... ...

This README is obsolete. It is included for historical purposes only. =================================================================== SIP User Agent README ==================================================================== Vovida SIP User Agent (ua) Release 1.5.0 alpha 1 October 30, 2002 ==================================================================== LICENSE AND COPYRIGHT ==================================================================== For information about licensing, please see the LICENSE file in this directory, or http://www.vovida.org/license.html . ==================================================================== INTRODUCTION ==================================================================== IMPORTANT: This SIP User Agent has been supersceded by SIPSet. SIPSet is in the sip/gua directory (rather than the sip/ua directory). As a result, this user agent is no longer being updated. Please see the sip/gua directory instead. This directory contains source code for the SIP User Agent (ua). The SIP User Agent is an implementation of the Vovida SIP stack that supports SIP call initiation, setup, and teardown. You can make a basic SIP call between two SIP user agents or from a user agent to a proxy server. ==================================================================== NEW FEATURES AND FUNCTIONALITY IN THIS RELEASE ==================================================================== Changes since 1.3.0: -- Persistent TCP connection support. -- Limited NAT traversal. -- New Linux sound card device. Changes since 1.2.0: -- Added support for direct entry of a URL to dial to Sound Card, NullHardware and QuickNet devices in Linux. Note that direct URL entry does not work in Windows. This release implements these new features and functionality: -- Support for Open Sound System (OSS) compatible sound cards -- Improved registration support -- A multitude of bug fixes ==================================================================== KNOWN BUGS OR LIMITATIONS ==================================================================== These bugs or limitations are known to exist: 1. Due to API changes within the sip/base source code, some of the ua's functionality has been compromised. Things that have been broken prior to this release are: -- consultation and blind transfer -- call waiting 2. The ua assumes the codec type is always PCM u-law. 3. The ua does not handle message retransmit timeout. 4. The ua does not do complete SDP negotiation. 5. There is a known problem when using the INTERNAL_IP_DIAL(3) speed dial option. Workaround: Use the new more flexible URL dialing. 6. There is no real implementation of single-line hold. If Ua_Xfer_Mode is set to 'Off', then a flashhook during a call will have no effect. Setting this parameter to 'Transfer' will put the far end on hold, but the local ua will then play a dialtone and wait for a second connection to be initiated. A flashhook or onhook at this point will take the initial call off hold, but this is different than stand-alone hold functionality. 7. Waiting too long at the URL dial prompt causes the call to time out. Workaround: Increase the Initial_Digit_Timeout and Inter_Digit_Timeout timer in configuration (ua.cfg) file. 8. Direct URL entry does not work in Windows and requires restarting the ua to reset the console input. 9. Early media (18x with SDP) does not work with QuickNet device and requires restarting the ua to reset the card. ==================================================================== FIXED LIMITATIONS ==================================================================== These limitations have been fixed: Changes since 1.3.0: -- UA now will respond to a 407 on a REGISTER or an INVITE. It should also respond to a 401 on a REGISTER or an INVITE, but that have not been tested. -- Suppress SIP SNMP agent in WIN32. -- Change size of WIN32 dummy include files to non-zero for Winzip. -- Eliminate unnecessary error message when turn on a two way RTP session by turing on the receiving part first. Changes since 1.2.0: -- The SDP message in the ACK message is now checked. -- QuickNet device is working again RedHat 7.1 and RedHat 7.2. ==================================================================== OTHER IMPLEMENTATION ISSUES ==================================================================== Please see the FAQ for more information. ==================================================================== GETTING STARTED ==================================================================== PLATFORM SUPPORT -------------------------------------------------------------------- The SIP User Agent will compile and run on Linux and Solaris operating systems. We have tested the SIP User Agent on: -- RedHat Linux 6.2 with gcc 2.91.66 (egcs-1.1.2 release) and glibc 2.1.3 -- RedHat Linux 7.1, 7.2 and 7.3 with gcc 2.96 -- Sun Solaris with gcc/g++ 2.96 compiler or Workshop 6 update 1 -- Windows with MSVC++ 6.0, Service Pack 4 The SIP User Agent has been compiled and tested on other distributions in the past, and should be distribution independent. Please consult the SIP mailing list as other developers may have tested the SIP stack and User Agent on other Linux distributions. SOUND CARD SUPPORT -------------------------------------------------------------------- Open Sound System (OSS) compatible sound cards should work with the ua. We have tested the SIP User Agent with Creative Labs SoundBlaster Live! sound card using the emu10k1 driver available at http://opensource.creative.com/ QUICKNET SUPPORT -------------------------------------------------------------------- The SIP User Agent can be used with the following Quicknet Cards: -- Internet PhoneJACK Card -- Internet LineJACK Card -- Internet PhoneCARD (PCMCIA) You can connect an analog phone to Internet PhoneJACK or LineJACK card and initiate a call from the analog phone. The ua requires the 1.0.0 version of the Quicknet drivers. The drivers, and instructions for installing them, are available from http://www.openh323.org/driver/build.html To use the Internet PhoneCard (PCMCIA), connect a headset (microphone/headset) and use the keyboard to initiate, answer, or hangup a call. Quicknet Driver Support ----------------------- In order to use the Quicknet cards in a Linux machine, you must install Quicknet Linux drivers. We have used the 1.0.0 drivers, from http://www.openh323.org/driver/build.html In our tests, the drivers which come with Redhat 7.1 do NOT work. Please contact RedHat with support issues with kernels from them, or upgrade to the 1.0.0 drivers as outlined on the web page above. COMPILING THE USER AGENT -------------------------------------------------------------------- To compile the user agent, type: make ==================================================================== USING THE USER AGENT ==================================================================== You can make a simple SIP call between two User Agents. There are two options supported: -- SIP signalling without voice path -- SIP signalling with voice path -- using non-PMCIA Quicknet Card -- using sound card or Quicknet Internet PhoneCard (PMCIA) If you want to test SIP signalling without voice (Null Hardware Option): -------------------------------------------------------------------- 1. Install and compile the SIP stack and SIP User Agent on each machine. 2. Copy and modify the ua.cfg file for each User Agent. 3. Run the User Agent by typing: ua -f ua.cfg. Refer to the "User Agent Options" section for more information on other options. 3. Press the 'a' key to go offhook (lift handset). 4. Using the keypad, press the digits keys to dial a number. Or, enter a SIP URL. 5. Press the 'z' key to go onhook (hangup). If you want to make a voice call with non-PMCIA Quicknet Cards: -------------------------------------------------------------------- 1. Install and compile the SIP stack and SIP User Agent on each machine. 2. Install a Quicknet card and Linux Quicknet drivers on each machine. Connect your phone to the Quicknet card. 3. Copy and Modify the ua.cfg file for each User Agent. 4. Run the User Agent by typing: ua -q -f ua.cfg. Refer to the "User Agent Options" section for more information on other options. 5. Using the phone, dial a number. Or, using the keyboard, enter a SIP URL. If you want to make a voice call using a sound card or the Quicknet Internet PhoneCARD (PMCIA): ------------------------------------------------------------------- 1. Install and compile the SIP stack and SIP User Agent on each machine. 2. Install the Quicknet Internet PhoneCARD (PMCIA) or sound card. 3. Attach the headset/microphone to your Quicknet card or sound card. 3. Copy and modify the ua.cfg file for each User Agent 4. Run the User Agent by typing: ua -s -f ua.cfg. Refer to the "User Agent Options" section for more information on other options, including the -s option for sound cards. 5. Press the 'a' key to go offhook (lift handset). 6. Using the keypad, press the digits keys to dial a number. Or, enter a SIP URL. 7. Press the 'z' key to go onhook (hangup). MODIFYING THE UA CONFIGURATION FILE -- ./SampleConfigFiles/ua.cfg -------------------------------------------------------------------- The ua.cfg is a configuration file for the User Agent. You must copy and modify the ua.cfg for each ua. A sample configuration file, ua.cfg, is include in the ua/SampleConfigFiles/ directory. The ua.cfg file provides comments that will help you modify the file. To make a simple User Agent to User Agent call without a proxy server, registration, authentication, or other features modify these fields: 1. Specify the a phone number for the User Agent in User_Name. 2. Add a # before Proxy_Server to disable this option. 3. Turn off registration by changing the Register_On field to False. 4. Add the proper entries to the Speed Dial section, or use the new direct entry of a URL. USER AGENT OPTIONS ------------------------------------------------------------------- To use the User Agent type: ./ua [options] where options are: -d Run in daemon mode (DO NOT USE.) -h Prints "ua [-dhqr] [-v[]] [-f ]" -q Use a Quicknet card for speech path. -s Use an OSS driver enabled sound card for speech path. Note: -s & -q are mutually exclusive. -r Run with retransmission of SIP messages off (Note. This option is not used. Currently retransmission is always off. DO NOT USE.) -f Configuration file -v[] This option allows you to operate in debug mode and debug comments will appear on your screen. The most commonly used log levels are LOG_DEBUG and LOG_DEBUG_STACK. When emailing the SIP list with questions about the ua, we suggest including a ua log using -vLOG_DEBUG_STACK. Generally, "debug mode" indicates a level of LOG_DEBUG or LOG_DEBUG_STACK. Levels: - LOG_EMERG - LOG_ALERT - LOG_CRIT - LOG_ERR - LOG_WARNING (default) - LOG_NOTICE - LOG_INFO - LOG_DEBUG - LOG_DEBUG_STACK Note: that there is no space between -v and the log level. For example, -vLOG_DEBUG, would turn on application level debug messages. QuickNet Card DIAL OPTIONS ------------------------------------------------------------------- When initiating a call using your keyboard, this option is supported after the phone is off-hook: Key Action --- ------ u Direct entry of a SIP URL NULL HARDWARE and Sound Card DIAL OPTIONS ------------------------------------------------------------------- When initiating or receiving a call using your keyboard, these options are supported: Key Action --- ------ a Go offhook (lift handset) z Go onhook (hangup) f Flashook u Direct entry of a SIP URL 1-9,#,* Equivalent buttons on the phone. DIAL TONES ------------------------------------------------------------------- Dial tones and ring back tones are generated locally using tone files. To enable these tones, copy the directory ua/Tone into the same directory from where you will run the ua. The directory must have proper read permissions. ==================================================================== CONTRIBUTORS ==================================================================== This software consists of voluntary contributions made by Vovida Networks, Inc., Cisco Systems, Inc., and many individuals including: Luan Dang Jeff Gao Cullent Jennings Sunitha Kumar Mahesh Shankar Mitch Zollinger ==================================================================== CONTACT INFORMATION AND WEBSITE ==================================================================== We welcome your feedback, suggestions and contributions. Contact us via email if you have questions, feedback, code submissions, and bug reports. For general inquiries - info@vovida.org We have mailing lists for the VOCAL applications and protocol stacks: VOCAL - vocal@vovida.org COPS - cops@vovida.org MGCP - mgcp@vovida.org RADIUS - radius@vovida.org RTP - rtp@vovida.org SIP - sip@vovida.org TRIP - trip@vovida.org You can subscribe to the mailing lists on www.vovida.org. ====================================================================== Copyright 2000-2003, Cisco Systems, Inc. THE INFORMATION HEREIN IS PROVIDED ON AN "AS IS" BASIS, WITHOUT ANY WARRANTIES OR REPRESENTATIONS, EXPRESS, IMPLIED OR STATUTORY, INCLUDING WITHOUT LIMITATION, WARRANTIES OF NONINFRINGEMENT, MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. $Id$

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