文件列表:

Trtsp-101tarh\rtsp-1.0.1\vocal\build\CVS\Entries (546, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\CVS\Repository (20, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\CVS\Root (37, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\global.h (2813, 2001-06-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\LICENSE (2563, 2000-10-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.all (4204, 2001-05-13)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.conf.in (279, 2001-05-09)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.opt (2920, 2001-05-13)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.osarch (4824, 2001-07-04)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.pkg (18111, 2001-05-19)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.post (12833, 2001-07-13)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.pre (4684, 2001-05-13)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\Makefile.tools (2210, 2001-06-29)
Trtsp-101tarh\rtsp-1.0.1\vocal\build\vovida-endian.h (586, 2000-05-03)
Trtsp-101tarh\rtsp-1.0.1\vocal\configure (29045, 2001-05-09)
Trtsp-101tarh\rtsp-1.0.1\vocal\configure.in (1038, 2001-05-09)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\CVS\Entries (44, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\CVS\Entries.Log (151, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\CVS\Repository (22, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\CVS\Root (37, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\CVS\Entries (337, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\CVS\Repository (34, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\CVS\Root (37, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\ftw.c (7511, 2000-08-23)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\ftw.h (1434, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\getopt.h (2863, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\getopt_long.3 (8281, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\getopt_long.c (12708, 2001-04-10)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\libcext_bsd.h (347, 2000-10-03)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_bsd\Makefile (115, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\CVS\Entries (436, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\CVS\Repository (35, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\CVS\Root (37, 2001-07-27)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\gethostby_r.c (3921, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\gethostby_r.o (2016, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\getnetby_r.c (2279, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\getprotoby_r.c (2318, 2000-08-22)
Trtsp-101tarh\rtsp-1.0.1\vocal\contrib\libcext_lgpl\getpw_r.c (2356, 2000-08-22)
... ...

==================================================================== VOCAL Readme ==================================================================== Software release number: VOCAL 1.3.0-RC1 Software release date: July 19, 2001 Readme release version: 1.3.0-RC1 Readme release date: July 19, 2001 ==================================================================== LICENSE AND COPYRIGHT ==================================================================== The Vovida Software License, Version 1.0 Copyright (c) 2000 Vovida Networks, Inc. All rights reserved. Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: 1. Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. 2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. 3. The names "VOCAL", "Vovida Open Communication Application Library", and "Vovida Open Communication Application Library (VOCAL)" must not be used to endorse or promote products derived from this software without prior written permission. For written permission, please contact VOCAL@vovida.org. 4. Products derived from this software may not be called "VOCAL", nor may "VOCAL" appear in their name, without prior written permission of Vovida Networks, Inc. THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT ARE DISCLAIMED. IN NO EVENT SHALL VOVIDA NETWORKS, INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT DAMAGES IN EXCESS OF $1,000, NOR FOR ANY INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -------------------------------------------------------------------- All third party licenses and copyright notices and other required legends also need to be complied with as well. ==================================================================== INTRODUCTION ==================================================================== The Vovida Open Communication Application Library (VOCAL) is an open source project targeted at speeding the adoption of Voice over Internet Protocol (VoIP) implementations by helping developers within the community build VoIP features, applications and services. The VOCAL software includes a Session Initiation Protocol- (SIP) based Redirect Server (RS), Feature Server (FS), Provisioning Server (PS), Policy Server (PoS) and Marshal Server (MS), along with protocol translators from SIP to H.323 and SIP to Media Gateway Control Protocol (MGCP). See the Documentation section of http://www.vovida.org for more information about VOCAL. ==================================================================== ERRATA ==================================================================== For the latest breaking news regarding this release candidate, please see the VOCAL Errata section in Faq-o-matic, at http://www.vovida.org/cgi-bin/fom?file=519 Issues and more information will be available from this location. ==================================================================== NEW FEATURES AND FUNCTIONS IN THIS RELEASE ==================================================================== FIXME: As we add more RCs, please move changes from the "since last RC" section to the "since 1.2.0" section. Changes since 1.3.0-RC1: ------------------------ --VThreadPool Bug: Fix for crash bug in VThreadPool. This problem was a subtle one appearing in all versions of VThreadPool, possibly causing hard-to-track crashes, particularly on Solaris with the pserver. Changes since 1.3.0-RC0: ------------------------ ---Deploy: There is a new allinone make target. Users should now be able to do: make all make allinone and have a basic working all in one system (with two users -- 1000 and 1001) built and installed into /usr/local/vocal. Alternately, users can do: make all make staging_notar # change to root make allinone if they do not wish to compile as root. "make allinone" does not replace deploy for deploying multiuser systems, although the intent is to eventually replace it. --Directory Structure change: The VOCAL make system stores its objects and binaries in a subdirectory relative to the code, named obj.{objecttype}.{arch} and bin.{objecttype}.{arch}, respectively, where {objecttype} is the object type (typically "debug"), and {arch} is the machine type (such as sun4u for sun sparc boxes, and i686 for intel machines). These directories have been renamed to obj.{objecttype}.{os}.{arch} and bin.{objecttype}.{os}.{arch}. In the new scheme, os stands in for the operating system, such as SunOS, Linux, or FreeBSD. Doing this makes it easier to build VOCAL for multiple operating systems when mounted over NFS. ---TCP Support: TCP support has been enhanced, thanks to contributions submitted by members of the Vovida.org community. ---UA Marshal Server: The UAMS responds to SIP CANCEL requests with a 200, OK response. Before, the UAMS was not handling the CANCEL message correctly. Simplified view of the old scenario: UA1 UAMS UA2 |---------CANCEL--------->|----------CANCEL--------->| | |<---------200-------------| |---------CANCEL--------->| | |---------CANCEL--------->| | |---------CANCEL--------->| | |---------CANCEL--------->| | Simplified view of the new scenario: UA1 UAMS UA2 |---------CANCEL--------->| | |<---------200------------| | | |----------CANCEL--------->| | |<---------200-------------| ---H.323 Translator: The H.323 Translator now supports gateway trunking. Before, it was only supporting NetMeeting as endpoints. Changes since 1.2.0: -------------------- ---Platform: Redhat 7.0 and 7.1 / Solaris 8 with g++ Support This RC contains support for compilation under Redhat 7.0 and 7.1, as well as Solaris 8 using g++ as the C++ compiler. Please see below for further details. In order to compile with g++ under Solaris 8, you must run the configure script from within the vocal directory: ./configure --with-toolchain=gnu --with-ar=/usr/local/bin/ar after this, you may use "make all" to compile the software. WARNING: You MUST have GNU ar to correctly build using gcc / g++ . The ar which comes with Solaris (in /usr/ccs/bin/ar) will not work -- it will crash with a segmentation fault if you fail to pass a valid location for GNU ar. WARNING: DO NOT mix objects compiled with Forte with objects compiled with g++. Doing so may result in compiler errors, as Forte is not binary compatible with g++. ---Performance: 500 Calls Per Second On Linux RedHat 6.2, performance ratings of 40 calls per second (cps) have been achieved over a minimum system consisting of 2 Marshal Servers(MS's) and 1 Redirect Server(RS). This performance is scalable and has been demonstrated processing up to 500 cps over 26 MS's, 4 RS's, and a Provisioning Server. This marks a four-fold improvement over vocal 1.2.0. During performance testing, a call was defined as follows: 1. 1 SIP endpoint called another SIP endpoint 2. The called endpoint answered the call 3. The called endpoint held the call for two seconds 4. The calling endpoint tore down the call. For more information about the performance testing, see the VOCAL Performance Specification Document: http://www.vovida.org/document/pdf/perfNumber.pdf ---SIP Stack: Enhanced TCP Support The new release candidate, vocal 1.3.0-RC0 offers more reliable and stable support for SIP over TCP than vocal 1.2.0. ==================================================================== BUG FIXES ==================================================================== Please refer to the Bugzilla listings on the http://www.vovida.org web site for bug fix information. ==================================================================== KNOWN LIMITATIONS ==================================================================== Known limitations for this RC: This RC has not undergone a full test of the system components, but only a basic call through the traditional (rs/ms/fs) system. FIXME: The following limitations are from VOCAL 1.2.0. They will be updated in a later RC for VOCAL 1.3.0. ---Provisioning Server (PS) Limitation: On the Solaris platform, loading VOCAL onto boxes with dual cpu's forces the Provisioning Server to crash. This problem has not appeared in our testing on Linux RedHat versions 6.2 and 7.1. Workaround: Disable one of the cpu's. Limitation: On the Solaris platform, loading more than 40,000 users makes the Provisioning GUI turn completely white and, therefore, practically unusable. Workaround: Do not load more than 40,000 users onto a VOCAL system running on Solaris. Limitation: If the user has the wrong file permissions for provisioning data, the data becomes corrupted. Workaround: Make sure that the user who runs the PS has write permissions for all of the provisioning data files. ---Feature Servers (FS) Limitation: Call Processing Language (CPL) scripts that are not generated by the VOCAL PS are not supported. Workaround: None. Limitation: Call Return, Call Waiting, Call Hold and Consultation Transfer do not work with Caller ID Blocking. Workaround: Disable Caller ID Blocking to make these features work. Limitation: Call Blocking only works with North American Dialing Plan numbers. Workaround: None. Limitation: If the administrator blocks 1+ calls, 1-8xx calls are purposely unblocked. This includes toll-free numbers as well as calls to ordinary 8xx area codes, such as 801 and 802. Workaround: None. Limitation: Phone numbers entered for call screening must include the area code regardless if they are local or long- distance numbers. Workaround: None. ---Redirect Server (RS) Limitation: Only one registration is stored per subscriber. When another registration comes in, the old information is overwritten. Workaround: None. Limitation: Registration Expire is currently turned off. When a subscriber expires, its terminating contacts are not removed from the subscriber database. Workaround: Delete the lines (332 and 334) before and after the clearTerm() function call (line 333) in Subscriber.cxx. ---Marshal Servers (MS) Limitation: Unknown SIP methods are not proxied. Workaround: None. Limitation: Conference calls, both meet-me and ad-hoc, require the inclusion of a conference bridge (a.k.a. Multimedia Conference Unit (MCU)) from an outside vendor to perform the audio mixing. Workaround: None. Limitation: If the MCU provides a SIP interface, conference calls can be placed to it through any type of MS. However, if the MCU provides a PSTN interface, the only way to route calls to it is through this path: a Gateway MS to a SIP-PSTN gateway to the MCU. When setting up the conference MS, use the system dialing plan to direct calls to the MCU; the gateway setting is ignored. Limitation: Each Gateway, Internetwork, and Conference MS supports one and only one external entity, whether that entity is a gateway, a proxy server, or a conference bridge. Workaround: Run a MS for each external entity except for the User Agent Marshal Servers (UAMS). Limitation: Conference calling between VOCAL UA endpoints is not reliable. Workaround: Restart the Conference MS and VOCAL UA endpoints. ---Policy Server Limitation: The VOCAL 1.2.0 Policy Server supports Open Settlement Protocol (OSP) version 1.4.3 only. Workaround: Run Voice Mail User Agents in Linux systems. ---H.323 Endpoint Limitation: Calls between H.323 and MGCP endpoints do not work. Workaround: None. ---Simple Network Management Protocol (SNMP) Limitation: On Solaris, if the system is not deployed, the port on which the SNMP daemon listens to the agents, port 33602, has to be free. Workaround: If this port is being used by a different application, kill that application before restarting snmpd. Limitation: Multi-Router Traffic Grapher (MRTG) is not supported in the VOCAL 1.2.0 release. It will be released in the next release patch. ---Voice Mail Limitation: VOCAL 1.2.0 doesn't support voice mail on Solaris. Workaround: Run Voice Mail User Agents in Linux systems. Limitation: The Voice Mail Server (VMS) always listens to 8024 port on Linux. Workaround: If this port is being used by a different application, kill that application before restarting the VMS. ---Media Gateway Control Protocol (MGCP) Endpoint Limitation: During calls between MGCP/UA's, consultation transfer to the PSTN cannot be completed. Workaround: None. Limitation: During calls from Cisco 7960 IP Phones to MGCP/H.323 endpoints, hanging up MGCP/H.323 endpoint does not clear the call completely on the Cisco 7960 IP Phone. Workaround: Hang up the Cisco 7960 IP Phone. Limitation: Conference and Consultation Transfer involving Telogy is not supported in VOCAL 1.2.0 due to limitations with the Telogy gateway. Workaround: Make conference and consultation transfers between three MGCP endpoints. ---Call Detail Record (CDR) Server Limitation: VOCAL 1.2.0 on Solaris doesn't connect to Remote Authentication Dial-In User Service- (RADIUS) based billing systems. Workaround: Use billing.dat files which are generated by the CDR Server to retrieve billing data. ---Java Telephony Application Programming Interface (JTAPI) Server Limitation: Once a VOCAL UA is provisioned with a JTAPI feature, the user cannot dial from their phone set without bringing up a JTAPI client. Workaround: Disable JTAPI feature for the user if the feature is not needed or bring up a JTAPI Client. ---User Agent (UA) Limitation: Consultation and Blind Transfer do not work Workaround: It is a bug. See Bugzilla for information about the fix. Limitation: Call Waiting does not work Workaround: None Limitation: VOCAL 1.2.0 does not support the load generator. Workaround: None. Limitation: The UA assumes that the codec type is always PCM U-law. Workaround: None. Limitation: The UA does not handle message retransmit timeouts. Workaround: None. Limitation: There is a known problem when using the INTERNAL_IP_DIAL(3) speed dial option. The INTERNAL_IP_DIAL speed dial plan is currently set to this dial pattern: DIAL_PATTERN string 3 ^[0-9][0-9][0-9]. The INTERNAL_IP_DIAL option will not work if the host portion of the IP address contains only 1 or 2 digits. For example, if you attempt to call 192.168.121.10 by dialing 10, the UA will take the previous digit and dial 110. Workaround: If the IP address contains 1 digit, dial "00" plus the digit. For example, if the last octect is "1", dial "001". If the address contains 2 digits, dial "0" plus the digits. For example, if the last octet is "11", dial "011". ==================================================================== GETTING STARTED ==================================================================== TESTED PLATFORMS -------------------------------------------------------------------- The software has been compiled using the following operating systems / versions: Operating System Kernel Compiler Linux (Redhat 6.2) 2.2.14 g++ 2.91.66 Linux (Redhat 7.1) 2.4 g++ 2.96 (Redhat) Solaris 8 2.8 sun workshop 6 update 1 Solaris 8 2.8 g++ 2.95.2 Java We recommend running the Java Client using the Java Plug-in version 1.3 or higher. Use Internet Explorer version 5 or higher, or use Netscape version 4.7 or higher. You can download the latest version of the Java Plug-in from http://www.javasoft.com/j2se/1.3/jre/index.html. TESTED GATEWAYS AND ENDPOINTS -------------------------------------------------------------------- FIXME: This section reflects the tested equipment for 1.2.0, and will be updated for 1.3.0 on a later RC. Cisco 7960 SIP IP Phone: Firmware version P0S3Z444 is currently in Beta trials and has been thouroughly tested with VOCAL 1.2.0. Firmware version P0S30200 is currently available and has undergone basic testing with VOCAL 1.2.0. MGCP gateway: Cisco IOS version 12.1(5)XM 2600 gateway: Cisco IOS version 12.1(5)XM 5300 gateway: Cisco IOS version 12.1(5)XM INSTALLATION and COMPILE INSTRUCTIONS -------------------------------------------------------------------- ---Single Host Installation and Compile Instructions: On a linux box either Redhat 6.2 or Redhat 7.1 do: 1. tar xvzf vocal-1.3.0-RC1.tar.gz 2. cd vocal-1.3.0-RC1 3. make all /* remove ldap2user from all: in vocal/Makefile on RH 6.2 */ 4. Install JRE 1.3 or equivalent jdk /* as root, #rpm -hivv jre*rpm */ 5. Install apache /*as root, #rpm -hivv apach*rpm */ 6 make allinone /* as root */ /* choose everything as default except /usr/java/jre1.3.1/bin/java, if you used jre. Might be different if you used jdk */ 7. /etc/rc.d/init.d/httpd restart /* as root */ /* note restart, not start */ 8 Export NPX_PLUGIN_PATH=/usr/java/jre1.3.1/plugins/i386/ns4 /* start netscape-communicator& from this xterm */ 9. On the browser enter http://hostname/vocal/ /* note the / after vocal */ 10. Copy ua1000.cfg and ua1001.cfg from vocal/sip/ua/Sample*/ua100?.cfg to /usr/local/vocal/bin 11. Edit ua100?.cfg by changing vocal-ext to the name of your host machine /* all inonebox hostname */ 12. You can run ./ua -r -f ./ua1000.cfg on one xterm and ./ua -rs -f ./ua1001.cfg on another xterm 13. Press 'a' for offhook, 'z' for on hook on the ua term. Dial 1000 or 1001 to call the other phone. ---Distrib ... ...

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