iptel.org SIP Express Router v0.11.0 -- Admin's Guide Jiri Kuthan Jan Janak Yacine Rebahi Copyright © 2001, 2002 FhG Fokus The document describes the SIP Express Router and its use in SIP networks. It is intended as an aid to server administrators. This documentation is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA For more details see the file COPYING in the source distribution of SER. _________________________________________________________ Table of Contents 1. General Information 1.1. About SIP Express Router (SER) 1.2. About iptel.org 1.3. Feature List 1.4. Use Cases 1.4.1. Added-Value ISP Services 1.4.2. PC2Phone 1.4.3. PBX Replacement 1.5. About SIP Technology 1.6. Known SER Limitations 1.7. Licensing 1.8. Obtaining Technical Assistance 1.9. More Information 1.10. Release Notes 2. Introduction to SER 2.1. Request Routing and SER Scripts 2.2. Conditional Statements 2.2.1. Operators and Operands 2.2.2. URI Matching 2.3. Request URI Rewriting 2.4. Destination Set 2.5. User Location 2.6. External Modules 2.7. Writing Scripts 2.7.1. Default Configuration Script 2.7.2. Stateful User Agent Server 2.7.3. Redirect Server 2.7.4. Executing External Script 2.7.5. On-Reply Processing (Forward on Unavailable) 3. Server Operation 3.1. Recommended Operational Practices 3.2. HOWTOs 3.2.1. User Management 3.2.2. User Aliases 3.2.3. Access Control (PSTN Gateway) 3.2.4. Accounting 3.2.5. Reliability 3.2.6. Stateful versus Stateless Forwarding 3.2.7. Serving Multiple Domains 3.2.8. Reporting Missed Calls 3.2.9. NAT Traversal 3.2.10. Using Only Latest User's Contact for Forwarding 3.2.11. Authentication Policy: Prevention of Unauthorized Domain Name Use in From and More 3.2.12. Connecting to PBX Voicemail Using a Cisco Gateway 3.3. Troubleshooting 4. Application Writing 4.1. Using exec Module 4.2. Application FIFO Server 4.2.1. Advanced Example: Click-To-Dial 5. Complementary Applications 5.1. serctl command-line tool 5.2. Web User Provisioning -- serweb 5.3. Voicemail 5.3.1. Introduction 5.3.2. Advantages 5.3.3. Technical limitations 5.3.4. Compilation and installation 5.3.5. Example ser Config File 5.3.6. Availabilty, report bugs, contact the author 6. Reference 6.1. Core Options 6.2. Core Commands 6.3. Command Line Parameters 6.4. Modules 6.5. FIFO Commands Reference 6.6. Used Database Tables List of Tables 2-1. Valid Combinations of Operands and Operators in Expressions 2-2. URI-rewriting Using Built-In Actions 6-1. Frequently Used Module Actions 6-2. FIFO Commands List of Examples 2-1. Static Forwarding 2-2. Conditional Statement 2-3. Use of search Action in Conditional Expression 2-4. More examples of use of ser operators and operands in conditional statements 2-5. Use of uri==myself Expression 2-6. Domain Matching Using Regular Expressions 2-7. A simple Numbering Plan 2-8. Rewriting URIs 2-9. Rewriting URIs Using User Location Database 2-10. URI-rewriting Exercise 2-11. REGISTER Request 2-12. Use of serctl Tool to Query User Location 2-13. Use of User Location Actions 2-14. Using Modules 2-15. Parameters in built-in and exported actions 2-16. Module Parameters 2-17. Default Configuration Script 2-18. Stateful UA Server 2-19. Redirect Server 2-20. Executing External Script 2-21. On-Reply Processing 3-1. Using ngrep 3-2. Use of SIPSak for Learning SIP Path 3-3. serctl ps command 3-4. IP Address Comparison 3-5. Logging Script 3-6. "Routing-history" labels 3-7. Configuration of Use of Aliases 3-8. Script for Gateway Access Control 3-9. Configuration with Enabled Accounting 3-10. Script for Replication of User Contacts 3-11. Forwarding to PBX/Voicemail via Cisco Gateways 3-12. Processing of Loose Routes Must be Present 4-1. Using exec: Step 1 4-2. Using exec: Step 2, Who Called Me 4-3. Using exec: step 3, Make The Script Work For Anyone 4-4. Adding Stateful Processing 4-5. Full Example of exec Use 4-6. Use of serctl to Access FIFO Server 4-7. uptime FIFO Request 4-8. FIFO Errors 4-9. Showing User Contacts Using serctl 4-10. Sending IM From Shell Script 4-11. Manipulation of User Contacts 4-12. Call-Flow for Click-To-Dial Using REFER 4-13. Running the CTD Example 5-1. serctl usage 5-2. Example Output of Server Watching Command sc monitor 5-3. Example ser Config File 6-1. route 6-2. failure_route 6-3. Use of if 6-4. Use of if-else 6-5. isflagset 6-6. Use of append_branch 6-7. Use of len_gt _________________________________________________________ Chapter 1. General Information 1.1. About SIP Express Router (SER) SIP Express Router (SER) is an industrial-strength, free VoIP server based on the Session Initiation Protocol (SIP, RFC3261). It is engineered to power IP telephony infrastructures up to large scale. The server keeps track of users, sets up VoIP sessions, relays instant messages and creates space for new plug-in applications. Its proven interoperability guarantees seamless integration with components from other vendors, eliminating the risk of a single-vendor trap. It has successfully participated in various interoperability tests in which it worked with the products of other leading SIP vendors. The SIP Express Router enables a flexible plug-in model for new applications: Third parties can easily link their plug-ins with the server code and provide thereby advanced and customized services. In this way, plug-ins such as RADIUS accounting, SMS gateway, ENUM queries, or presence agent have already been developed and are provided as advanced features. Other modules are underway: firewall control, postgress and LDAP database drivers and more. Its performance and robustness allows it to serve millions of users and accommodate needs of very large operators. With a $3000 dual-CPU PC, the SIP Express Router is able to power IP telephony services in an area as large as the Bay Area during peak hours. Even on an IPAQ PDA, the server withstands 150 calls per second (CPS)! The server has been powering our iptel.org free SIP site withstanding heavy daily load that is further increasing with the popularity of Microsoft's Windows Messenger. The SIP Express Router is extremely configurable to allow the creation of various routing and admission policies as well as setting up new and customized services. Its configurability allows it to serve many roles: network security barrier, application server, or PSTN gateway guard for example. ser can be also used with contributed applications. Currently, serweb, a ser web interface, SIPSak diagnostic tool and SEMS media server are available. Visit our site, http://www.iptel.org/, for more information on contributed packages. _________________________________________________________ 1.2. About iptel.org iptel.org is a know-how organization spun off from Germany's national research company FhG Fokus. One of the first SIP implementations ever, low-QoS enhancements, interoperability tests and VoIP-capable firewall control concepts are examples of well-known FhG's work. iptel.org continues to keep this know-how leadership in SIP. The access rate of the company's site, a well-known source of technological information, is a best proof of interest. Thousands of hits come every day from the whole Internet. The iptel.org site, powered by SER, offers SIP services on the public Internet. Feel free to apply for a free SIP account at http://www.iptel.org/user/ _________________________________________________________ 1.3. Feature List Based on the latest standards, the SIP Express Router (SER) includes support for registrar, proxy and redirect mode. Further it acts as an application server with support for instant messaging and presence including a 2G/SMS and Jabber gateway, a call control policy language, call number translation, private dial plans and accounting, ENUM, authorization and authentication (AAA) services. SER runs on Sun/Solaris, PC/Linux, PC/BSD, IPAQ/Linux platforms and supports both IPv4 and IPv6. Hosting multiple domains and database redundancy is supported. ser has been carefully engineered with the following design objectives in mind: * Speed - With ser, thousands of calls per seconds are achievable even on low-cost platforms. This competitive capacity allows setting up networks which are inexpensive and easy to manage due to low number of devices required. The processing capacity makes dealing with many stress factors easier. The stress factors may include but are not limited to broken configurations and implementations, boot avalanches on power-up, high-traffic applications such as presence, redundancy replications and denial-of-service attacks. The speed has been achieved by extensive code optimization, use of customized code, ANSI C combined with assembly instructions and leveraging latest SIP improvements. When powered by a dual-CPU Linux PC, ser is able to process thousands of calls per second, capacity needed to serve call signaling demands of Bay Area population. * Flexibility - SER allows its users to define its behavior. Administrators may write textual scripts which determine SIP routing decisions, the main job of a proxy server. They may use the script to configure numerous parameters and introduce additional logic. For example, the scripts can determine for which destinations record routing should be performed, who will be authenticated, which transactions should be processed statefully, which requests will be proxied or redirected, etc. * Extensibility - SER's extensibility allows linking of new C code to ser to redefine or extend its logic. The new code can be developed independently on SER core and linked to it in run-time. The concept is similar to the module concept known for example in Apache Web server. Even such essential parts such as transaction management have been developed as modules to keep the SER core compact and fast. * Portability. ser has been written in ANSI C. It has been extensively tested on PC/Linux and Sun/Solaris. Ports to BSD and IPAQ/Linux exist. * Interoperability. ser is based on the open SIP standard. It has undergone extensive tests with products of other vendors both in iptel.org labs and in the SIP Interoperability Tests (SIPIT). ser powers the public iptel.org site 24 hours a day, 356 days a year serving numerous SIP implementations using this site. * Small size. Footprint of the core is 300k, add-on modules take up to 630k. _________________________________________________________ 1.4. Use Cases This section illustrates the most frequent uses of SIP. In all these scenarios, the SIP Express Router (SER) can be easily deployed as the glue connecting all SIP components together, be it soft-phones, hard-phones, PSTN gateways or any other SIP-compliant devices. _________________________________________________________ 1.4.1. Added-Value ISP Services To attract customers, ISPs frequently offer applications bundled with IP access. With SIP, the providers can conveniently offer a variety of services running on top of a single infrastructure. Particularly, deploying VoIP and instant messaging and presence services is as easy as setting up a SIP server and guiding customers to use Windows Messenger. Additionally, the ISPs may offer advanced services such as PSTN termination, user-driven call handling or unified messaging all using the same infrastructure. SIP Express Router has been engineered to power large scale networks: its capacity can deal with large number of customers under high load caused by modern applications. Premium performance allows deploying a low number of boxes while keeping investments and operational expenses extremely low. ISPs can offer SIP-based instant messaging services and interface them to other instant messaging systems (Jabber, SMS). VoIP can be easily integrated along with added-value services, such as voicemail. _________________________________________________________ 1.4.2. PC2Phone Internet Telephony Service Providers (ITSPs) offer the service of interconnecting Internet telephony users using PC softphone or appliances to PSTN. Particularly with long-distance and international calls, competitive pricing can be achieved by routing the calls over the Internet. SIP Express Router can be easily configured to serve pc2phone users, distribute calls to geographically appropriate PSTN gateway, act as a security barrier and keep track of charging. _________________________________________________________ 1.4.3. PBX Replacement Replacing a traditional PBX in an enterprise can achieve reasonable savings. Enterprises can deploy a single infrastructure for both voice and data and bridge distant locations over the Internet. Additionally, they can benefit of integration of voice and data. The SIP Express Router scales from SOHOs to large, international enterprises. Even a single installation on a common PC is able to serve VoIP signaling of any world's enterprise. Its policy-based routing language makes implementation of numbering plans of companies spread across the world very easy. ACL features allow for protection of PSTN gateway from unauthorized callers. SIP Express Router's support for programmable routing and accounting efficiently allows for implementation of such a scenario. _________________________________________________________ 1.5. About SIP Technology The SIP protocol family is the technology which integrates services. With SIP, Internet users can easily contact each other; figure out willingness to have a conversation and couple different applications such as VoIP, video and instant messaging. Integration with added-value services is seamless and easy. Examples include integration with web (click-to-dial), E-mail (voice2email, UMS), and PSTN-like services (conditional forwarding). The core piece of the technology is the Session Initiation Protocol (SIP, RFC3261) standardized by IETF. Its main function is to establish communication sessions between users connected to the public Internet and identified by e-mail-like addresses. One of SIP's greatest features is its transparent support for multiple applications: the same infrastructure may be used for voice, video, gaming or instant messaging as well as any other communication application. There are numerous scenarios in which SIP is already deployed: PBX replacement allows for deployment of single inexpensive infrastructure in enterprises; PC-2-phone long-distance services (e.g., Deltathree) cut callers long-distance expenses; instant messaging offered by public severs (e.g., iptel.org) combines voice and text services with presence information. New deployment scenarios are underway: SIP is a part of UMTS networks and research publications suggest the use of SIP for virtual home environments or distributed network games. _________________________________________________________ 1.6. Known SER Limitations The following items are not part of current distribution and are planned for next releases: * Script processing of multiple branches on forking Warning ser's request processing language allows to make request decisions based on current URI. When a request if forked to multiple destinations, only the first branch's URI is used as input for script processing. This might lead to unexpected results. Whenever a URI resolves to multiple different next-hop URIs, only the first is processed which may result in handling not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP address and PSTN gateway SIP address. If the IP phone address is the first, then script execution ignores the second branch. If a script includes checking gateway address in request URI, the checks never match. That might result in ignoring of gateway admission control rules or applying them unnecessarily to non-gateway destinations. List of known problems is publicly available at the ser webpage at http://www.iptel.org/ser/ . See the "ISSUES" link. _________________________________________________________ 1.7. Licensing ser is freely available under terms and conditions of the GNU General Public License. ----------------------------------------------------------------------- -- IMPORTANT NOTES 1) The GPL applies to this copy of SIP Express Router software (ser). For a license to use the ser software under conditions other than those described here, or to purchase support for this software, please contact iptel.org by e-mail at the following addres ses: info@iptel.org (see http://www.gnu.org/copyleft/gpl-faq.html#TOCHeardOtherLicense for an explanation how parallel licenses comply with GPL) 2) ser software allows programmers to plug-in external modules to the core part. Note that GPL mandates all plug-ins developed for the ser software released under GPL license to be GPL-ed as well. (see http://www.gnu.org/copyleft/gpl-faq.html#GPLAndPlugins for a detailed explanation) 3) Note that the GPL bellow is copyrighted by the Free Software Foundat ion, but the ser software is copyrighted by FhG ----------------------------------------------------------------------- -- GNU Licence FAQ This FAQ provides answers to most frequently asked questions. To fully understand implications of the GNU license, read it. - you can run SER for any purpose - you can redistribute it as long as you include source code and license conditions with the distribution - you cannot release programs derived from SER without releasing their source code ----------------------------------------------------------------------- -- GNU GENERAL PUBLIC LICENSE Version 2, June 1991 Copyright (C) 1989, 1991 Free Software Foundation, Inc. 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. Preamble The licenses for most software are designed to take away your freedom to share and change it. By contrast, the GNU General Public License is intended to guarantee your freedom to share and change free software--to make sure the software is free for all its users. This General Public License applies to most of the Free Software Foundation's software and to any other program whose authors commit to using it. (Some other Free Software Foundation software is covered by the GNU Library General Public License instead.) You can apply it to your programs, too. When we speak of free software, we are referring to freedom, not price. Our General Public Licenses are designed to make sure that you have the freedom to distribute copies of free software (and charge for this service if you wish), that you receive source code or can get it if you want it, that you can change the software or use pieces of it in new free programs; and that you know you can do these things. To protect your rights, we need to make restrictions that forbid anyone to deny you these rights or to ask you to surrender the rights. These restrictions translate to certain responsibilities for you if you distribute copies of the software, or if you modify it. For example, if you distribute copies of such a program, whether gratis or for a fee, you must give the recipients all the rights that you have. You must make sure that they, too, receive or can get the source code. And you must show them these terms so they know their rights. We protect your rights with two steps: (1) copyright the software, an d (2) offer you this license which gives you legal permission to copy, distribute and/or modify the software. Also, for each author's protection and ours, we want to make certain that everyone understands that there is no warranty for this free software. If the software is modified by someone else and passed on, w e want its recipients to know that what they have is not the original, so that any problems introduced by others will not reflect on the original authors' reputations. Finally, any free program is threatened constantly by software patents. We wish to avoid the danger that redistributors of a free program will individually obtain patent licenses, in effect making the program proprietary. To prevent this, we have made it clear that any patent must be licensed for everyone's free use or not licensed at all. The precise terms and conditions for copying, distribution and modification follow. GNU GENERAL PUBLIC LICENSE TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION 0. This License applies to any program or other work which contains a notice placed by the copyright holder saying it may be distributed under the terms of this General Public License. The "Program", below, refers to any such program or work, and a "work based on the Program" means either the Program or any derivative work under copyright law: that is to say, a work containing the Program or a portion of it, either verbatim or with modifications and/or translated into another language. (Hereinafter, translation is included without limitation in the term "modification".) Each licensee is addressed as "you". Activities other than copying, distribution and modification are not covered by this License; they are outside its scope. The act of running the Program is not restricted, and the output from the Program is covered only if its contents constitute a work based on the Program (independent of having been made by running the Program). Whether that is true depends on what the Program does. 1. You may copy and distribute verbatim copies of the Program's source code as you receive it, in any medium, provided that you conspicuously and appropriately publish on each copy an appropriate copyright notice and disclaimer of warranty; keep intact all the notices that refer to this License and to the absence of any warranty; and give any other recipients of the Program a copy of this License along with the Program. You may charge a fee for the physical act of transferring a copy, and you may at your option offer warranty protection in exchange for a fee. 2. You may modify your copy or copies of the Program or any portion of it, thus forming a work based on the Program, and copy and distribute such modifications or work under the terms of Section 1 above, provided that you also meet all of these conditions: a) You must cause the modified files to carry prominent notices stating that you changed the files and the date of any change. b) You must cause any work that you distribute or publish, that in whole or in part contains or is derived from the Program or any part thereof, to be licensed as a whole at no charge to all third parties under the terms of this License. c) If the modified program normally reads commands interactively when run, you must cause it, when started running for such interactive use in the most ordinary way, to print or display an announcement including an appropriate copyright notice and a notice that there is no warranty (or else, saying that you provide a warranty) and that users may redistribute the program under these conditions, and telling the user how to view a copy of this License. (Exception: if the Program itself is interactive but does not normally print such an announcement, your work based on the Program is not required to print an announcement.) These requirements apply to the modified work as a whole. If identifiable sections of that work are not derived from the Program, and can be reasonably considered independent and separate works in themselves, then this License, and its terms, do not apply to those sections when you distribute them as separate works. But when you distribute the same sections as part of a whole which is a work based on the Program, the distribution of the whole must be on the terms of this License, whose permissions for other licensees extend to the entire whole, and thus to each and every part regardless of who wrote i t. Thus, it is not the intent of this section to claim rights or contest your rights to work written entirely by you; rather, the intent is to exercise the right to control the distribution of derivative or collective works based on the Program. In addition, mere aggregation of another work not based on the Program with the Program (or with a work based on the Program) on a volume of a storage or distribution medium does not bring the other work under the scope of this License. 3. You may copy and distribute the Program (or a work based on it, under Section 2) in object code or executable form under the terms of Sections 1 and 2 above provided that you also do one of the following: a) Accompany it with the complete corresponding machine-readable source code, which must be distributed under the terms of Sections 1 and 2 above on a medium customarily used for software interchange ; or, b) Accompany it with a written offer, valid for at least three years, to give any third party, for a charge no more than your cost of physically performing source distribution, a complete machine-readable copy of the corresponding source code, to be distributed under the terms of Sections 1 and 2 above on a medium customarily used for software interchange; or, c) Accompany it with the information you received as to the offer to distribute corresponding source code. (This alternative is allowed only for noncommercial distribution and only if you received the program in object code or executable form with such an offer, in accord with Subsection b above.) The source code for a work means the preferred form of the work for making modifications to it. For an executable work, complete source code means all the source code for all modules it contains, plus any associated interface definition files, plus the scripts used to control compilation and installation of the executable. However, as a special exception, the source code distributed need not include anything that is normally distributed (in either source or binary form) with the major components (compiler, kernel, and so on) of the operating system on which the executable runs, unless that component itself accompanies the executable. If distribution of executable or object code is made by offering access to copy from a designated place, then offering equivalent access to copy the source code from the same place counts as distribution of the source code, even though third parties are not compelled to copy the source along with the object code. 4. You may not copy, modify, sublicense, or distribute the Program except as expressly provided under this License. Any attempt otherwise to copy, modify, sublicense or distribute the Program is void, and will automatically terminate your rights under this License. However, parties who have received copies, or rights, from you under this License will not have their licenses terminated so long as such parties remain in full compliance. 5. You are not required to accept this License, since you have not signed it. However, nothing else grants you permission to modify or distribute the Program or its derivative works. These actions are prohibited by law if you do not accept this License. Therefore, by modifying or distributing the Program (or any work based on the Program), you indicate your acceptance of this License to do so, and all its terms and conditions for copying, distributing or modifying the Program or works based on it. 6. Each time you redistribute the Program (or any work based on the Program), the recipient automatically receives a license from the original licensor to copy, distribute or modify the Program subject to these terms and conditions. You may not impose any further restrictions on the recipients' exercise of the rights granted herein. You are not responsible for enforcing compliance by third parties to this License. 7. If, as a consequence of a court judgment or allegation of patent infringement or for any other reason (not limited to patent issues), conditions are imposed on you (whether by court order, agreement or otherwise) that contradict the conditions of this License, they do not excuse you from the conditions of this License. If you cannot distribute so as to satisfy simultaneously your obligations under this License and any other pertinent obligations, then as a consequence you may not distribute the Program at all. For example, if a patent license would not permit royalty-free redistribution of the Program by all those who receive copies directly or indirectly through you, then the only way you could satisfy both it and this License would be to refrain entirely from distribution of the Program. If any portion of this section is held invalid or unenforceable under any particular circumstance, the balance of the section is intended to apply and the section as a whole is intended to apply in other circumstances. It is not the purpose of this section to induce you to infringe any patents or other property right claims or to contest validity of any such claims; this section has the sole purpose of protecting the integrity of the free software distribution system, which is implemented by public license practices. Many people have made generous contributions to the wide range of software distributed through that system in reliance on consistent application of that system; it is up to the author/donor to decide if he or she is willing to distribute software through any other system and a licensee cannot impose that choice. This section is intended to make thoroughly clear what is believed to be a consequence of the rest of this License. 8. If the distribution and/or use of the Program is restricted in certain countries either by patents or by copyrighted interfaces, the original copyright holder who places the Program under this License may add an explicit geographical distribution limitation excluding those countries, so that distribution is permitted only in or among countries not thus excluded. In such case, this License incorporates the limitation as if written in the body of this License. 9. The Free Software Foundation may publish revised and/or new versio ns of the General Public License from time to time. Such new versions wil l be similar in spirit to the present version, but may differ in detail t o address new problems or concerns. Each version is given a distinguishing version number. If the Program specifies a version number of this License which applies to it and "any later version", you have the option of following the terms and conditio ns either of that version or of any later version published by the Free Software Foundation. If the Program does not specify a version number of this License, you may choose any version ever published by the Free Sof tware Foundation. 10. If you wish to incorporate parts of the Program into other free programs whose distribution conditions are different, write to the auth or to ask for permission. For software which is copyrighted by the Free Software Foundation, write to the Free Software Foundation; we sometime s make exceptions for this. Our decision will be guided by the two goals of preserving the free status of all derivatives of our free software a nd of promoting the sharing and reuse of software generally. NO WARRANTY 11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRA NTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHE N OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRES SED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING , REPAIR OR CORRECTION. 12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRI TING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAG ES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARI SING OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMIT ED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED B Y YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OT HER PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. END OF TERMS AND CONDITIONS _________________________________________________________ 1.8. Obtaining Technical Assistance iptel.org offers qualified professional services. We help you to plan your network, configure your server, build applications, integrate SIP components with each other, and set up advanced features such as redundancy, multidomain support, CLID interworking and others not described in this document. Our customer alert services notifies you on all new features and code fixes. We help you to solve operational troubles in short time and keep you updated on latest operational practices. Ask info@iptel.org for information on enrollment in our support program. Additionaly, help may be obtained from our user forum. The community of SER users is subscribed to the serusers@iptel.org mailing list and discusses issues related to SER operation. Mailing List Instructions * Public archives and subscription form: http://mail.iptel.org/mailman/listinfo/serusers * To post, send an email to serusers@iptel.org * If you think you encountered an error, please submit the following information to avoid unnecessary round-trip times: + Name and version of your operating system -- you can obtain it by calling uname -a + ser distribution: release number and package + ser build -- you can obtain it by calling ser -V + Your ser configuration file + ser logs -- with default settings few logs are printed to syslog facility which typically dumps them to /var/log/messages. To enable detailed logs dumped to stderr, apply the following configuration options: debug=8, log_stderror=yes, fork=no. + Captured SIP messages -- you can obtain them using tools such as ngrep or ethereal. If you are concerned about your privacy and do not wish your queries to be posted and archived publicly, you may post to serhelp@iptel.org. E-mails to this address are only forwarded to iptel.org's ser development team. However, as the team is quite busy you should not be surprised to get replies with considerable delay. _________________________________________________________ 1.9. More Information Most up-to-date information including latest and most complete version of this documentation is always available at our website, http://www.iptel.org/ser/. The site includes links to other important information about ser, such as installation guidelines (INSTALL), download links, development pages, programmer's manual, etc. A SIP tutorial (slide set) is available at http://www.iptel.org/sip/ . _________________________________________________________ 1.10. Release Notes Release notes for SIP Express Router (ser) *********************************************** $Id: NEWS,v 1.15.2.1 2003/08/27 07:57:08 calrissian Exp $ *********************************************** * Changes introduced in 0.8.11 *********************************************** +--------------------------------------------------------+ | CAUTION: the 0.8.11 release include changes which | | are incompatible with scripts and databases used | | in previous versions. Care is advised when upgrading | | from previous releases to 0.8.11. | +--------------------------------------------------------+ New features ============= - RFC3261 support - TCP support and cross-transport forwarding [core] - loose routing support [rr module] - New modules - vm -- voicemail interface [vm] - ENUM support [enum] - presence agent [pa] - dynamic domain management -- allows to manipulate hosting of multiple domains in run-time [module] - flat-text-file database support [dbtext] - rich access control lists [permissions] - Feature Improvements - click-to-dial, which is based on improved tm/FIFO that better supports external applications [tm module] - web accounting -- acc module can report to serweb on placed calls [acc module] - improved exec module (header fields passed now as environment variables to scripts) [exec module] - Architectural Improvements - powerpc fast locking support - netbsd support - 64 bits arch. support (e.g. netbsd/sparc64). - New Experimental Features (not tested at all yet) - nathelper utility for Cisco/ATA NAT traversal [nathelper] - another NAT traversal utility [mangler] - postgress support [postgress] - pdt module (prefix2domain) [pdt] Changes to use of ser scripts ============================= About Multiple Transport Support -------------------------------- SER now suports multiple transport protocols: UDP and TCP. As there may be UAs which support only either protocol and cannot speak to each other directly, we recommend to alway record-route SIP re quests, to keep the transport-translating SER in path. Also, if a dest ination transport is not known, stateful forwarding is recommended -- use of stateless forwarding for TCP2UDP would result in loss of relia bility. core ---- - reply_route has been renamed to failure_route -- the old nam e caused too much confusion - forward_tcp and forward_udp can force SER to forward via spe cific transport protocol acc module: ----------- - radius and sql support integrated in this module; you need t o recompile to enable it - acc_flag is now called log_flag to better reflect it relates to the syslog mode (as opposed to sql/radius); for the same reasons, the accounting action is now called "acc_log_reques t" and the option for missed calls "log_missed_calls" - log_fmt allows now to specify what will be printed to syslog auth module: ------------ - auth module has been split in auth, auth_db, auth_radius, gr oup group_radius, uri and uri_radius - all the parameters that were part of former auth module are now part of auth_db module - auth_db module contains all functions needed for database authentication - auth_radius contains functions needed for radius authenticat ion - group module contains group membership checking functions - group_radius contains radius group membeship checking functi ons - is_in_group has been renamed to is_user_in and places to gro ups module - check_to and check_from have been moved to the uri module im module: ---------- - im is no longer used and has been obsoleted by TM exec module: ------------ - exec_uri and exec_user have been obsoleted by exec_dset; exec_dset is identical to exec_uri in capabilities; it additionaly passes content of request elements (header fields and URI parts) in environment variables; users of exec_user can use exec_dset now and use the "URI_USER" variable to learn user part of URI - exec_dset and exec_msg return false, if return value of script does not euqal zero - exec_dset takes an additional parameter, which enables validation of SIP URIs returned by external application jabber module: -------------- - presence support for Jabber users is enabled loading the PA module and using handle_subscribe("jabber") for SUBSCRIBE requests to jabber user msilo module: ------------- - m_store has now a parameter to set what should be considered for storing as destination uri. This enables support for sav ing the messages on negative replies. radius_acc module: ------------------ - radius_acc module has been removed and radius accounting is now part of acc module registrar/usrloc modules: ------------------------- - multi domain support, the modules user username@domain as AO R if enabled - descent modification time ordering of contacts - case sensitive/insensitive comparison of URI can be enabled rr module: ---------- - addRecordRoute has been replaced with record_route - rewriteFromRoute has been replaced with loose_route() - a new option, "enable_full_lr" can be set to make life with misimplemented UAs easier and put LR in from "lr=on" - rr module can insert two Record-Route header fields when necesarry (disconnected networks, UDP->TCP and so on) tm module: ---------- - t_reply_unsafe, used in former versions within reply_routes, is deprecated; now t_reply is used from any places in script - t_on_negative is renamed to t_on_failure -- the old name jus t caused too much confusion - FIFO t_uac used by some applications (like serweb) has been replaced with t_uac_dlg (which allows easier use by dialog- oriented applications, like click-to-dial) - if you wish to do forward to another destination from failure_route (reply_route formerly), you need to call t_rel ay or t_relay_to explicitely now - t_relay_to has been replaced with t_relay_to_udp and t_relay _to_tcp _________________________________________________________ Chapter 2. Introduction to SER 2.1. Request Routing and SER Scripts The most important concept of every SIP server is that of request routing. The request routing logic determines the next hop of a request. It can be for example used to implement user location service or enforce static routing to a gateway. Real-world deployments actually ask for quite complex routing logic, which needs to reflect static routes to PSTN gateways, dynamic routes to registered users, authentication policy, capabilities of SIP devices, etc. SER's answer to this need for routing flexibility is a routing language, which allows administrators to define the SIP request processing logic in a detailed manner. They can for example easily split SIP traffic by method or destination, perform user location, trigger authentication, verify access permissions, and so on. The primary building block of the routing language are actions. There are built-in actions (like forward for stateless forwarding or strip for stripping URIs) as well as external actions imported from shared library modules. All actions can be combined in compound actions by enclosing them in braces, e.g. {a1(); a2();}. Actions are aggregated in one or more route blocks. Initially, only the default routing block denoted by route[0] is called. Other routing blocks can be called by the action route(blocknumber), recursion is permitted. The language includes conditional statements. The routing script is executed for every received request in sequential order. Actions may return positive/negative/zero value. Positive values are considered success and evaluated as TRUE in conditional expressions. Negative values are considered FALSE. Zero value means error and leaves execution of currently processed route block. The route block is left too, if break is explicitly called from it. The easiest and still very useful way for ser users to affect request routing logic is to determine next hop statically. An example is routing to a PSTN gateway whose static IP address is well known. To configure static routing, simply use the action forward( IP_address, port_number). This action forwards an incoming request "as is" to the destination described in action's parameters. Example 2-1. Static Forwarding # if requests URI is numerical and starts with # zero, forward statelessly to a static destination if (uri=~"^sip:0[0-9]*@iptel.org") { forward( 192.168.99.3, 5080 ); } However, static forwarding is not sufficient in many cases. Users desire mobility and change their location frequently. Lowering costs for termination of calls in PSTN requires locating a least-cost gateway. Which next-hop is taken may depend on user's preferences. These and many other scenarios need the routing logic to be more dynamic. We describe in Section 2.2 how to make request processing subject to various conditions and in Section 2.3 how to determine next SIP hop. _________________________________________________________ 2.2. Conditional Statements A very useful feature is the ability to make routing logic depend on a condition. A script condition may for example distinguish between request processing for served and foreign domains, IP and PSTN routes, it may split traffic by method or username, it may determine whether a request should be authenticated or not, etc. ser allows administrators to form conditions based on properties of processed request, such as method or uri, as well as on virtually any piece of data on the Internet. Example 2-2. Conditional Statement This example shows how a conditional statement is used to split incoming requests between a PSTN gateway and a user location server based on request URI. # if request URI is numerical, forward the request to PSTN gateway... if (uri=~"^sip:[0-9]+@foo.bar") { # match using a regular expression forward( gateway.foo.bar, 5060 ); } else { # ... forward the request to user location server otherwise forward( userloc.foo.bar, 5060 ); }; Conditional statements in ser scripts may depend on a variety of expressions. The simplest expressions are action calls. They return true if they completed successfully or false otherwise. An example of an action frequently used in conditional statements is search imported from textops module. search action leverages textual nature of SIP and compares SIP requests against a regular expression. The action returns true if the expression matched, false otherwise. Example 2-3. Use of search Action in Conditional Expression # prevent strangers from claiming to belong to our domain; # if sender claims to be in our domain in From header field, # better authenticate him if (search("(f|From): .*@mydomain.com)) { if (!(proxy_authorize("mydomain.com" /* realm */,"subscriber" /* ta ble name */ ))) { proxy_challenge("mydomain.com /* ream */, "1" /* use qop */ ); break; } } As modules may be created, which export new functions, there is virtually no limitation on what functionality ser conditions are based on. Implementers may introduce new actions whose return status depends on request content or any external data as well. Such actions can query SQL, web, local file systems or any other place which can provide information wanted for request processing. Furthermore, many request properties may be examined using existing built-in operands and operators. Available left-hand-side operands and legal combination with operators and right-hand-side operands are described in Table 2-1. Expressions may be grouped together using logical operators: negation (!), AND (&&), OR ( || and precedence parentheses (()). _________________________________________________________ 2.2.1. Operators and Operands There is a set of predefined operators and operands in ser, which in addition to actions may be evaluated in conditional expressions. Left hand-side operands, which ser understands are the following: * method, which refers to request method such as REGISTER or INVITE * uri, which refers to current request URI, such as "sip:john.doe@foo.bar" Note Note that "uri" always refers to current value of URI, which is subject to change be uri-rewriting actions. * src_ip, which refers to IP address from which a request came. Warning Note that comparison of src_ip to an IP address may cause DNS lookups and delay request processing. To avoid DNS lookups, don't enclose IP addresses in quotes. Otherwise, reverse DNS lookup can be performed to compare to host aliases. * dst_ip refers to server's IP address at which a request was received * src_port port number from which a SIP request came ser understands the following operators: * == stands for equity * =~ stands for regular expression matching * logical operators: and, or, negation, parentheses (C-notation for the operators may be used too) Table 2-1. Valid Combinations of Operands and Operators in Expressions left-hand-side operand valid operators valid right-hand side operators examples/comments method == (exact match), =~ (regular expression matching) string method=="INVITE" || method=="ACK" || method=="CANCEL" uri == (exact match), =~ (regular expression matching) string uri=="sip:foo@bar.com" matches only if exactly this uri is in request URI == (exact match) myself the expression uri==myself is true if the host part in request URI equals a server name or a server alias (set using the alias option in configuration file) src_ip == (match) IP, IP/mask_length, IP/mask, hostname, myself src_ip==192.168.0.0/16 matches requests coming from a private network dst_ip == (match) IP, IP/mask_length, IP/mask, hostname, myself dst_ip==127.0.0.1 matches if a request was received via loopback interface src_port == (match) port number port number from which a request was sent, e.g. src_port==5060 Example 2-4. More examples of use of ser operators and operands in conditional statements # using an action as condition input; in this # case, an actions 'search' looks for Contacts # with private IP address in requests; the condition # is processed if such a contact header field is # found if (search("^(Contact|m): .*@(192\.168\.|10\.|172\.16)")) { # .... # this condition is true if request URI matches # the regular expression "@bat\.iptel\.org" if (uri=~"@bat\.iptel\.org") { # ... # and this condition is true if a request came # from an IP address (useful for example for # authentication by IP address if digest is not # supported) AND the request method is INVITE # if ( (src_ip==192.68.77.110 and method=="INVITE") # ... _________________________________________________________ 2.2.2. URI Matching URI matching expressions have a broad use in a SIP server and deserve more explanation. Typical uses of URI matching include implementation of numbering plans, domain matching, binding external applications to specific URIs, etc. This section shows examples of typical applications of URI-matching. _________________________________________________________ 2.2.2.1. Domain Matching One of most important uses of URI matching is deciding whether a request is targeted to a served or outside domain. Typically, different request processing applies. Requests for outside domains are simply forwarded to them, whereas more complex logic applies to requests for a served domain. The logic may include saving user's contacts when REGISTER requests are received, forwarding requests to current user's location or a PSTN gateways, interaction with external applications, etc. The easiest way to decide whether a request belongs a served domain is using the myself operand. The expression "uri==myself" returns true if domain name in request URI matches name of the host at which ser is running. This may be insufficient in cases when server name is not equal to domain name for which the server is responsible. For example, the "uri==myself" condition does not match if a server "sipserver.foo.bar" receives a request for "sip:john.doe@foo.bar". To match other names in URI than server's own, set up the alias configuration option. The option may be used multiple times, each its use adds a new item to a list of aliases. The myself condition returns then true also for any hostname on the list of aliases. Example 2-5. Use of uri==myself Expression # ser powers a domain "foo.bar" and runs at host sipserver.foo.bar; # Names of served domains need to be stated in the aliases # option; myself would not match them otherwise and would only # match requests with "sipserver.foo.bar" in request-URI alias="foo.bar" alias="sales.foo.bar" route[0] { if (uri==myself) { # the request either has server name or some of the # aliases in its URI log(1,"request for served domain") # some domain-specific logic follows here .... } else { # aha -- the server is not responsible for this # requests; that happens for example with the following URI s # - sip:a@marketing.foo.bar # - sip:a@otherdomain.bar log(1,"request for outbound domain"); # outbound forwarding t_relay(); }; } It is possible to recognize whether a request belongs to a domain using regular expressions too. Care needs to be paid to construction of regular expressions. URI syntax is rich and an incorrect expression would result in incorrect call processing. The following example shows how an expression for domain matching can be formed. Example 2-6. Domain Matching Using Regular Expressions In this example, server named "sip.foo.bar" with IP address 192.168.0.10 is responsible for the "foo.bar" domain. That means, requests with the following hostnames in URI should be matched: * foo.bar, which is the name of server domain * sip.foo.bar, since it is server's name and some devices put server's name in request URI * 192.168.0.10, since it is server's IP address and some devices put server's IP address in request URI Note how this regular expression is constructed. In particular: * User name is optional (it is for example never included in REGISTER requests) and there are no restrictions on what characters it contains. That is what (.+@)? mandates. * Hostname must be followed by port number, parameters or headers -- that is what the delimiters [:;\?] are good for. If none it these follows, the URI must be ended ($). Otherwise, longer hostnames such as 192.168.0.101 or foo.bar.otherdomain.com would mistakenly match. * Matches are case-insensitive. All hostnames "foo.bar", "FOO.BAR" and "FoO.bAr" match. if (uri=~"^sip:(.+@)?(192\.168\.0\.10|(sip\.)?foo\.bar)([:;\?].*)?$") log(1, "yes, it is a request for our domain"); break; }; _________________________________________________________ 2.2.2.2. Numbering Plans Other use of URI matching is implementation of dialing plans. A typical task when designing a dialing plan for SIP networks is to distinguish between "pure-IP" and PSTN destinations. IP users typically have either alphanumerical or numerical usernames. The numerical usernames are convenient for PSTN callers who can only use numeric keypads. Next-hop destination of IP users is looked up dynamically using user location database. On the other hand, PSTN destinations are always indicated by nummerical usernames. Requests to PSTN are statically forwarded to well-known PSTN gateways. Example 2-7. A simple Numbering Plan This example shows a simple dialing plan which reserves dialing prefix "8" for IP users, other numbers are used for PSTN destinations and all other non-nummerical usernames are used for IP users. # is it a PSTN destination? (is username nummerical and does not begin with 8?) if (uri=~"^sip:[0-79][0-9]*@") { # ... forward to gateways then; # check first to which PSTN destination the requests goes; # if it is US (prefix "1"), use the gateway 192.168.0.1... if (uri=~"^sip:1") { # strip the leading "1" strip(1); forward(192.168.0.1, 5060); } else { # ... use the gateway 10.0.0.1 for all other destinations forward(10.0.0.1, 5060); } break; } else { # it is an IP destination -- try to lookup it up in user location DB if (!lookup("location")) { # bad luck ... user off-line sl_send_reply("404", "Not Found"); break; } # user on-line...forward to his current destination forward(uri:host,uri:port); } _________________________________________________________ 2.3. Request URI Rewriting The ability to give users and services a unique name using URI is a powerful tool. It allows users to advertise how to reach them, to state to whom they wish to communicate and what services they wish to use. Thus, the ability to change URIs is very important and is used for implementation of many services. "Unconditional forwarding" from user "boss" to user "secretary" is a typical example of application relying on change of URI address. ser has the ability to change request URI in many ways. A script can use any of the following built-in actions to change request URI or a part of it: rewriteuri, rewritehost, rewritehostport, rewriteuser, rewriteuserpass and rewriteport. When later in the script a forwarding action is encountered, the action forwards the request to address in the rewritten URI. Example 2-8. Rewriting URIs if (uri=~"dan@foo.bar") { rewriteuri("sip:bla@somewherelse.com") # forward statelessly to the destination in current URI, i.e., # to sip:bla@somewherelese.com:5060 forward( uri:host, uri:port); } Two more built-in URI-rewriting commands are of special importance for implementation of dialing plans and manipulation of dialing prefixes. prefix(s) , inserts a string "s" in front of SIP address and strip(n) takes away the first "n" characters of a SIP address. See Table 2-2 for examples of use of built-in URI-rewriting actions. Commands exported by external modules can change URI too and many do so. The most important application is changing URI using the user location database. The command lookup(table) looks up current user's location and rewrites user's address with it. If there is no registered contact, the command returns a negative value. Example 2-9. Rewriting URIs Using User Location Database # store user location if a REGISTER appears if (method=="REGISTER") { save("mydomain1"); } else { # try to use the previously registered contacts to # determine next hop if(lookup("mydomain1")) { # if found, forward there... t_relay(); } else { # ... if no contact on-line, tell it upstream sl_send_reply("404", "Not Found" ); }; }; External applications can be used to rewrite URI too. The "exec" module provides script actions, which start external programs and read new URI value from their output. exec_dset both calls an external program, passes SIP request elements to it, waits until it completes, and eventually rewrites current destination set with its output. It is important to realize that ser operates over current URI all the time. If an original URI is rewritten by a new one, the original will will be forgotten and the new one will be used in any further processing. In particular, the uri matching operand and the user location action lookup always take current URI as input, regardless what the original URI was. Table 2-2 shows how URI-rewriting actions affect an example URI, sip:12345@foo.bar:6060. Table 2-2. URI-rewriting Using Built-In Actions Example Action Resulting URI rewritehost("192.168.0.10") rewrites the hostname in URI, other parts (including port number) remain unaffected. sip:12345@192.168.10:6060 rewriteuri("sip:alice@foo.bar"); rewrites the whole URI completely. sip:alice@foo.bar rewritehostport("192.168.0.10:3040")rewrites both hostname and port number in URI. sip:12345@192.168.0.10:3040 rewriteuser("alice") rewrites user part of URI. sip:alice@foo.bar:6060 rewriteuserpass("alice:pw") replaces the pair user:password in URI with a new value. Rewriting password in URI is of historical meaning though, since basic password has been replaced with digest authentication. sip:alice:pw@foo.bar:6060 rewriteport("1234") replaces port number in URI sip:12345@foo.bar:1234 prefix("9") inserts a string ahead of user part of URI sip:912345@foo.bar:6060 strip(2) removes leading characters from user part of URI sip:345@foo.bar:6060 You can verify whether you understood URI processing by looking at the following example. It rewrites URI several times. The question is what is the final URI to which the script fill forward any incoming request. Example 2-10. URI-rewriting Exercise exec_dset("echo sip:2234@foo.bar; echo > /dev/null"); strip(2); if (uri=~"^sip:2") { prefix("0"); } else { prefix("1"); }; forward(uri:host, uri:port); The correct answer is the resulting URI will be "sip:134@foo.bar". exec_dset rewrites original URI to "sip:2234@foo.bar", strip(2) takes two leading characters from username away resulting in "34@iptel.org", the condition does not match because URI does not begin with "2" any more, so the prefix "1" is inserted. _________________________________________________________ 2.4. Destination Set Whereas needs of many scenarios can by accommodated by maintaining a single request URI, some scenarios are better served by multiple URIs. Consider for example a user with address john.doe@iptel.org. The user wishes to be reachable at his home phone, office phone, cell phone, softphone, etc. However, he still wishes to maintain a single public address on his business card. To enable such scenarios, ser allows translation of a single request URI into multiple outgoing URIs. The ability to forward a request to multiple destinations is known as forking in SIP language. All outogoing URIs (in trivial case one of them) are called destination set. The destination set always includes one default URI, to which additional URIs can be appended. Maximum size of a destination set is limited by a compile-time constant, MAX_BRANCHES, in config.h. Some actions are designed for use with a single URI whereas other actions work with the whole destination set. Actions which are currently available for creating the destination set are lookup from usrloc module and exec_dset from exec module. lookup fills in the destination set with user contact's registered previously with REGISTER requests. The exec actions fill in the destination set with output of an external program. In both cases, current destination set is completely rewritten. New URIs can be appended to destination set by a call to the built-in action append_branch(uri). Currently supported features which utilize destination sets are forking and redirection. Action t_relay (TM module) for stateful forwarding supports forking. If called with a non-trivial destination set, t_relay forks incoming request to all URIs in current destination set. See Example 2-9. If a user previously registered from three locations, the destination set is filled with all of them by lookup and the t_relay command forwards the incoming request to all these destinations. Eventually, all user's phone will be ringing in parallel. SIP redirection is another feature which leverages destination sets. It is a very light-weighted method to establish communication between two parties with minimum burden put on the server. In ser, the action sl_send_reply (SL module) is used for this purpose. This action allows to generate replies to SIP requests without keeping any state. If the status code passed to the action is 3xx, the current destination set is printed in reply's Contact header fields. Such a reply instructs the originating client to retry at these addresses. (See Example 2-19). Most other ser actions ignore destination sets: they either do not relate to URI processing ( log, for example) or they work only with the default URI. All URI-rewriting functions such as rewriteuri belong in this category. URI-comparison operands only refer to the first URI (see Section 2.2.1). Also, the built-in action for stateless forwarding, forward works only with the default URI and ignores rest of the destination set. The reason is a proxy server willing to fork must guarantee that the burden of processing multiple replies is not put unexpectedly on upstream client. This is only achievable with stateful processing. Forking cannot be used along with stateless forward, which thus only processes one URI out of the whole destination set. _________________________________________________________ 2.5. User Location Mobility is a key feature of SIP. Users are able to use one one or more SIP devices and be reachable at them. Incoming requests for users are forwarded to all user's devices in use. The key concept is that of soft-state registration. Users can -- if in possession of valid credentials -- link SIP devices to their e-mail like address of record. Their SIP devices do so using a REGISTER request, as in Example 2-11. The request creates a binding between the public address of record (To header field) and SIP device's current address (Contact header field). Example 2-11. REGISTER Request REGISTER sip:192.168.2.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.16;branch=z9hG4bKd5e5.5a9947e4.0 Via: SIP/2.0/UDP 192.168.2.33:5060 From: sip:123312@192.168.2.16 To: sip:123312@192.168.2.16 Call-ID: 00036bb9-0fd30217-491b6aa6-0a7092e9@192.168.2.33 Date: Wed, 29 Jan 2003 18:13:15 GMT CSeq: 101 REGISTER User-Agent: CSCO/4 Contact: sip:123312@192.168.2.33:5060 Content-Length: 0 Expires: 600 Similar requests can be used to query all user's current contacts or to delete them. All Contacts have certain time to live, when the time expires, contact is removed and no longer used for processing of incoming requests. ser is built to do both: update user location database from received REGISTER requests and look-up these contacts when inbound requests for a user arrive. To achieve high performance, the user location table is stored in memory. In regular intervals (usrloc module's parameter timer_interval determines their length), all changes to the in-memory table are backed up in mysql database to achieve peristence accross server reboots. Administrators or application writers can lookup list of current user's contacts stored in memory using the serctl tool (see Section 5.1). Example 2-12. Use of serctl Tool to Query User Location [jiri@fox jiri]$ sc ul show jiri ;q=0.00;expires=456 ;q=0.00;expires=36000 Building user location in ser scripts is quite easy. One first needs to determine whether a request is for served domain, as described in Section 2.2.2.1. If that is the case, the script needs to distinguish between REGISTER requests, that update user location table, and all other requests for which next hop is determined from the table. The save action is used to update user location (i.e., it writes to it). The lookup actions reads from the user location table and fills in destination set with current user's contacts. Example 2-13. Use of User Location Actions # is the request for my domain ? if (uri==myself) { if (method=="REGISTER") { # REGISTERs are used to update save("location"); break; # that's it, we saved the contacts, exit now } else { if (!lookup("location") { # no registered contact sl_send_reply("404", "Not Found"); break; } # ok -- there are some contacts for the user; forward # the incoming request to all of them t_relay(); }; }; Note that we used the action for stateful forwarding, t_relay. That's is because stateful forwarding allows to fork an incoming request to multiple destinations. If we used stateful forwarding, the request would be forwarded only to one uri out of all user's contacts. _________________________________________________________ 2.6. External Modules ser provides the ability to link the server with external third-party shared libraries. Lot of functionality which is included in the ser distribution is actually located in modules to keep the server "core" compact and clean. Among others, there are modules for checking max_forwards value in SIP requests (maxfwd), transactional processing (tm), record routing (rr), accounting (acc), authentication (auth), SMS gateway (sms), replying requests (sl), user location (usrloc, registrar) and more. In order to utilize new actions exported by a module, ser must first load it. To load a module, the directive loadmodule "filename" must be included in beginning of a ser script file. Example 2-14. Using Modules This example shows how a script instructs ser to load a module and use actions exported by it. Particularly, the sl module exports an action sl_send_reply which makes ser act as a stateless user agent and reply all incoming requests with 404. # first of all, load the module! loadmodule "/usr/lib/ser/modules/sl.so route{ # reply all requests with 404 sl_send_reply("404", "I am so sorry -- user not found"); } Note Note that unlike with core commands, all actions exported by modules must have parameters enclosed in quotation marks in current version of ser. In the following example, the built-in action forward for stateless forwarding takes IP address and port numbers as parameters without quotation marks whereas a module action t_relay for stateful forwarding takes parameters enclosed in quotation marks. Example 2-15. Parameters in built-in and exported actions # built-in action doesn't enclose IP addresses and port numbers # in quotation marks forward(192.168.99.100, 5060); # module-exported functions enclose all parameters in quotation # marks t_relay_to_udp("192.168.99.100", "5060"); Many modules also allow users to change the way how they work using predefined parameters. For example, the authentication module needs to know location of MySQL database which contains users' security credentials. How module parameters are set using the modparam directive is shown in Example 2-16. modparam always contains identification of module, parameter name and parameter value. Description of parameters available in modules is available in module documentation. Yet another thing to notice in this example is module dependency. Modules may depend on each other. For example, the authentication modules leverages the mysql module for accessing mysql databases and sl module for generating authentication challenges. We recommend that modules are loaded in dependency order to avoid ambiguous server behaviour. Example 2-16. Module Parameters # ------------------ module loading ---------------------------------- # load first modules on which 'auth' module depends; # sl is used for sending challenges, mysql for storage # of user credentials loadmodule "modules/sl/sl.so" loadmodule "modules/mysql/mysql.so" loadmodule "modules/auth/auth.so" # ------------------ module parameters ------------------------------- # tell the auth module the access data for SQL database: # username, password, hostname and database name modparam("auth", "db_url","sql://ser:secret@dbhost/ser") # ------------------------- request routing logic ------------------- # authenticate all requests prior to forwarding them route{ if (!proxy_authorize("foo.bar" /* realm */, "subscriber" /* table name */ )) { proxy_challenge("foo.bar", "0"); break; }; forward(192.168.0.10,5060); } _________________________________________________________ 2.7. Writing Scripts This section demonstrates simple examples how to configure server's behaviour using the ser request routing language. All configuration scripts follow the ser language syntax, which dictates the following section ordering: * global configuration parameters -- these value affect behaviour of the server such as port number at which it listens, number of spawned children processes, and log-level. See Section 6.1 for a list of available options. * module loading -- these statements link external modules, such as transaction management (tm) or stateless UA server (sl) dynamically. See Section 6.4 for a list of modules included in ser distribution. Note If modules depend on each other, than the depending modules must be loaded after modules on which they depend. We recommend to load first modules tm and sl because many other modules (authentication, user location, accounting, etc.) depend on these. * module-specific parameters -- determine how modules behave; for example, it is possible to configure database to be used by authentication module. * one or more route blocks containing the request processing logic, which includes built-in actions as well as actions exported by modules. See Section 6.2 for a list of built-in actions. * optionally, if modules supporting reply processing (currently only TM) are loaded, one or more failure_route blocks containing logic triggered by received replies. Restrictions on use of actions within failure_route blocks apply -- see Section 6.2 for more information. _________________________________________________________ 2.7.1. Default Configuration Script The configuration script, ser.cfg, is a part of every ser distribution and defines default behaviour. It allows users to register with the server and have requests proxied to each other. After performing routine checks, the script looks whether incoming request is for served domain. If so and the request is "REGISTER", ser acts as SIP registrar and updates database of user's contacts. Optionally, it verifies user's identity first to avoid unauthorized contact manipulation. Non-REGISTER requests for served domains are then processed using user location database. If a contact is found for requested URI, script execution proceeds to stateful forwarding, a negative 404 reply is generated otherwise. Requests outside served domain are always statefully forwarded. Note that this simple script features several limitations: * By default, authentication is turned off to avoid dependency on mysql. Unless it it turned on, anyone can register using any name and "steal" someone else's calls. * Even it authentication is turned on, there is no relationship between authentication username and address of record. That means that for example a user authenticating himself correctly with "john.doe" id may register contacts for "gw.bush". Site policy may wish to mandate authentication id to be equal to username claimed in To header field. check_to action from auth module can be used to enforce such a policy. * There is no dialing plan implemented. All users are supposed to be reachable via user location database. See Section 2.2.2.2 for more information. * The script assumes users will be using server's name as a part of their address of record. If users wish to use another name (domain name for example), this must be set using the alias options. See Section 2.2.2.1 for more information. * If authentication is turned on by uncommenting related configuration options, clear-text user passwords will by assumed in back-end database. Example 2-17. Default Configuration Script # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this confi g), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________ 2.7.2. Stateful User Agent Server This examples shows how to make ser act as a stateful user agent (UA). Ability to act as as a stateful UA is essential to many applications which terminate a SIP path. These applications wish to focus on their added value. They do not wish to be involved in all SIP gory details, such as request and reply retransmission, reply formatting, etc. For example, we use the UA functionality to shield SMS gateway and instant message store from SIP transactional processing. The simple example bellow issues a log report on receipt of a new transaction. If we did not use a stateful UA, every single request retransmission would cause the application to be re-executed which would result in duplicated SMS messages, instant message in message store or log reports. The most important actions are t_newtran and t_reply. t_newtran shields subsequent code from retransmissions. It returns success and continues when a new request arrived. It exits current route block immediately on receipt of a retransmission. It only returns a negative value when a serious error, such as lack of memory, occurs. t_reply generates a reply for a request. It generates the reply statefully, i.e., it is kept for future retransmissions in memory. Note Applications that do not need stateful processing may act as stateless UA Server too. They just use the sl_send_reply action to send replies to requests without keeping any state. The benefit is memory cannot run out, the drawback is that each retransmission needs to be processed as a new request. An example of use of a stateless server is shown in Section 2.7.3 and Section 2.7.4. Example 2-18. Stateful UA Server # # $Id: uas.cfg,v 1.7 2003/06/03 03:18:12 jiri Exp $ # # this example shows usage of ser as user agent # server which does some functionality (in this # example, 'log' is used to print a notification # on a new transaction) and behaves statefuly # (e.g., it retransmits replies on request # retransmissions) # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # create transaction state; abort if error occured if ( !t_newtran()) { sl_reply_error(); break; }; # the following log will be only printed on receipt of # a new message; retranmissions are absorbed by t_newtran log(1, "New Transaction Arrived\n"); # do what you want to do as a sever... if (uri=~"a@") { if (!t_reply("409", "Bizzar Error")) { sl_reply_error(); }; } else { if (!t_reply("699", "I don't want to chat with you")) { sl_reply_error(); }; }; } _________________________________________________________ 2.7.3. Redirect Server The redirect example shows how to redirect a request to multiple destination using 3xx reply. Redirecting requests as opposed to proxying them is essential to various scalability scenarios. Once a message is redirected, ser discards all related state and is no more involved in subsequent SIP transactions (unless the redirection addresses point to the same server again). The key ser actions in this example are append_branch and sl_send_reply (sl module). append_branch adds a new item to the destination set. The destinations set always includes the current URI and may be enhanced up to MAX_BRANCHES items. sl_send_reply command, if passed SIP reply code 3xx, takes all values in current destination set and adds them to Contact header field in the reply being sent. Example 2-19. Redirect Server # # $Id: redirect.cfg,v 1.5 2002/12/09 02:32:57 jiri Exp $ # # this example shows use of ser as stateless redirect server # # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@iptel.org:9"); # append one more URI to the destination ser append_branch("sip:redirect@iptel.org:9"); # redirect now sl_send_reply("300", "Redirect"); } _________________________________________________________ 2.7.4. Executing External Script Like in the previous example, we show how to make ser act as a redirect server. The difference is that we do not use redirection addresses hardwired in ser script but get them from external shell commands. We also use ser's ability to execute shell commands to log source IP address of incoming SIP requests. The new commands introduced in this example are exec_msg and exec_dset. exec_msg takes current requests, starts an external command, and passes the requests to the command's standard input. It also passes request's source IP address in environment variable named SRCIP. exec_dset serves for URI rewriting by external applications. The exec_dset action passes current URI to the called external program, and rewrites current destination set with the program's output. An example use would be an implementation of a Least-Cost-Router, software which returns URI of the cheapest PSTN provider for a given destination based on some pricing tables. Example 2-20 is much easier: it prints fixed URIs on its output using shell script echo command. Note This script works statelessly -- it uses this action for stateless replying, sl_send_reply. No transaction is kept in memory and each request retransmission is processed as a brand-new request. That may be a particular concern if the server logic (exec actions in this example) is too expensive. See Section 2.7.2 for instructions on how to make server logic stateful, so that retransmissions are absorbed and do not cause re-execution of the logic. Example 2-20. Executing External Script # # $Id: exec.cfg,v 1.7 2003/06/03 03:18:12 jiri Exp $ # # this example shows use of ser as stateless redirect server # which rewrites URIs using an exernal utility # # ------------------ module loading ---------------------------------- loadmodule "modules/exec/exec.so" loadmodule "modules/sl/sl.so" # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # first dump the message to a file using cat command exec_msg("printenv SRCIP > /tmp/exectest.txt; cat >> /tmp/exect est.txt"); # and then rewrite URI using external utility # note that the last echo command trashes input parameter if (exec_dset("echo sip:mra@iptel.org;echo sip:mrb@iptel.org;ec ho>/dev/null")) { sl_send_reply("300", "Redirect"); } else { sl_reply_error(); log(1, "alas, rewriting failed\n"); }; } _________________________________________________________ 2.7.5. On-Reply Processing (Forward on Unavailable) Many services depend on status of messages relayed downstream: forward on busy and forward on no reply to name the most well-known ones. To support implementation of such services, ser allows to return to request processing when request forwarding failed. When a request is reprocessed, new request branches may be initiated or the transaction can be completed at discretion of script writer. The primitives used are t_on_failure(r) and failure_route[r]{}. If t_on_failure is called before a request is statefuly forwarded and a forwarding failure occurs, ser will return to request processing in a failure_route block. Failures include receipt of a SIP error (status code >= 300 ) from downstream or not receiving any final reply within final response period. The length of the timer is governed by parameters of the tm module. fr_timer is the length of timer set for non-INVITE transactions and INVITE transactions for which no provisional response is received. If a timer hits, it indicates that a downstream server is unresponsive. fr_inv_timer governs time to wait for a final reply for an INVITE. It is typically longer than fr_timer because final reply may take long time until callee (finds a mobile phone in a pocket and) answers the call. In Example 2-21, failure_route[1] is set to be entered on error using the t_on_failure(1) action. Within this reply block, ser is instructed to initiate a new branch and try to reach called party at another destination (sip:nonsense@iptel.org). To deal with the case when neither the alternate destination succeeds, t_on_failure is set again. If the case really occurs, failure_route[2] is entered and a last resort destination (sip:foo@iptel.org) is tried. Example 2-21. On-Reply Processing # # $Id: onr.cfg,v 1.8 2003/06/03 03:18:12 jiri Exp $ # # example script showing both types of forking; # incoming message is forked in parallel to # 'nobody' and 'parallel', if no positive reply # appears with final_response timer, nonsense # is retried (serial forking); than, destination # 'foo' is given last chance # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" # ----------------- setting module-specific parameters --------------- # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # try these two destinations first in parallel; the second # destination is targeted to sink port -- that will make ser # wait until timer hits seturi("sip:nobody@iptel.org"); append_branch("sip:parallel@iptel.org:9"); # if we do not get a positive reply, continue at reply_route[1] t_on_failure("1"); # forward the request to all destinations in destination set no w t_relay(); } failure_route[1] { # forwarding failed -- try again at another destination append_branch("sip:nonsense@iptel.org"); log(1,"first redirection\n"); # if this alternative destination fails too, proceed to reply_r oute[2] t_on_failure("2"); t_relay(); } failure_route[2] { # try out the last resort destination append_branch("sip:foo@iptel.org"); log(1, "second redirection\n"); # we no more call t_on_negative here; if this destination # fails too, transaction will complete t_relay(); } _________________________________________________________ Chapter 3. Server Operation 3.1. Recommended Operational Practices Operation of a SIP server is not always easy task. Server administrators are challenged by broken or misconfigured user agents, network and host failures, hostile attacks and other stress-makers. All such situations may lead to an operational failure. It is sometimes very difficult to figure out the root reason of a failure, particularly in a distributed environment with many SIP components involved. In this section, we share some of our practices and refer to tools which have proven to make life of administrators easier 3.1.1. Keeping track of messages is good 3.1.2. Real-time Traffic Watching 3.1.3. Tracing Errors in Server Chains 3.1.4. Watching Server Health 3.1.5. Is Server Alive 3.1.6. Dealing with DNS 3.1.7. Logging 3.1.8. Labeling Outbound Requests 3.1.1. Keeping track of messages is good Frequently, operational errors are discovered or reported with a delay. Users frustrated by an error frequently approach administrators and scream "even though my SIP requests were absolutely ok yesterday, they were mistakenly denied by your server". If administrators do not record all SIP traffic at their site, they will be no more able to identify the problem reason. We thus recommend that site operators record all messages passing their site and keep them stored for some period of time. They may use utilities such as ngrep or tcpdump . There is also a utility scripts/harv_ser.sh in ser distribution for post-processing of captured messages. It summarizes messages captured by reply status and user-agent header field. 3.1.2. Real-time Traffic Watching Looking at SIP messages in real-time may help to gain understanding of problems. Though there are commercial tools available, using a simple, text-oriented tool such as ngrep makes the job very well thanks to SIP's textual nature. Example 3-1. Using ngrep In this example, all messages at port 5060 which include the string "bkraegelin" are captured and displayed [jiri@fox s]$ ngrep bkraegelin@ port 5060 interface: eth0 (195.37.77.96/255.255.255.240) filter: ip and ( port 5060 ) match: bkraegelin@ # U +0.000000 153.96.14.162:50240 -> 195.37.77.101:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 153.96.14.162:5060. From: sip:bkraegelin@iptel.org. To: sip:bkraegelin@iptel.org. Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162. Date: Thu, 26 Sep 2002 22:03:55 GMT. CSeq: 101 REGISTER. Expires: 10. Content-Length: 0. . # U +0.000406 195.37.77.101:5060 -> 153.96.14.162:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 153.96.14.162:5060. From: sip:bkraegelin@iptel.org. To: sip:bkraegelin@iptel.org. Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162. CSeq: 101 REGISTER. WWW-Authenticate: Digest realm="iptel.org", nonce="3d9385170000000043ac bf6ba9c9741790e0c57adee73812", algorithm=MD5. Server: Sip EXpress router(0.8.8 (i386/linux)). Content-Length: 0. Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=31604 req_src_ip =153.96.14.162 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==1". 3.1.3. Tracing Errors in Server Chains A request may pass any number of proxy servers on its path to its destination. If an error occurs in the chain, it is difficult for upstream troubleshooters and/or users complaining to administrators to learn more about error circumstances. ser does its best and displays extensive diagnostics information in SIP replies. It allows troubleshooters and/or users who report to troubleshooters to gain additional knowledge about request processing status. This extended debugging information is part of the warning header field. See Example 3-1 for an illustration of a reply that includes such a warning header field. The header field contains the following pieces of information: * Server's IP Address -- good to identify from which server in a chain the reply came. * Incoming and outgoing URIs -- good to learn for which URI the reply was generated, as it may be rewritten many times in the path. Particularly useful for debugging of numbering plans. * Number of Via header fields in replied request -- that helps in assessment of request path length. Upstream clients would not know otherwise, how far away in terms of SIP hops their requests were replied. * Server's process id. That is useful for debugging to discover situations when mutliple servers listen at the same address. * IP address of previous SIP hop as seen by the SIP server. If server administrator is not comfortable with disclosing all this information, he can turn them off using the sip_warning configuration option. A nice utility for debugging server chains is sipsak, SIP Swiss Army Knife, traceroute-like tool for SIP developed at iptel.org. It allows you to send OPTIONS request with low, increasing Max-Forwards header-fields and follow how it propagates in SIP network. See its webpage at http://sipsak.berlios.de/ . Example 3-2. Use of SIPSak for Learning SIP Path [jiri@bat sipsak]$ ./sipsak -T -s sip:7271@iptel.org warning: IP extract from warning activated to be more informational 0: 127.0.0.1 (0.456 ms) SIP/2.0 483 Too Many Hops 1: ?? (31.657 ms) SIP/2.0 200 OK without Contact header Note that in this example, the second hop server does not issue any warning header fields in replies and it is thus impossible to display its IP address in SIPsak's output. 3.1.4. Watching Server Health Watching Server's operation status in real-time may also be a great aid for trouble-shooting. ser has an excellent facility, a FIFO server, which allows UNIX tools to access server's internals. (It is similar to how Linux tool access Linux kernel via the proc file system.) The FIFO server accepts commands via a FIFO (named pipe) and returns data asked for. Administrators do not need to learn details of the FIFO communication and can serve themselves using a front-end utility serctl. Of particular interest for monitoring server's operation are serctl commands ps and moni. The former displays running ser processes, whereas the latter shows statistics. Example 3-3. serctl ps command This example shows 10 processes running at a host. The process 0, "attendant" watches child processes and terminates all of them if a failure occurs in any of them. Processes 1-4 listen at local interface and processes 5-8 listen at Ethernet interface at port number 5060. Process number 9 runs FIFO server, and process number 10 processes all server timeouts. [jiri@fox jiri]$ serctl ps 0 31590 attendant 1 31592 receiver child=0 sock=0 @ 127.0.0.1::5060 2 31595 receiver child=1 sock=0 @ 127.0.0.1::5060 3 31596 receiver child=2 sock=0 @ 127.0.0.1::5060 4 31597 receiver child=3 sock=0 @ 127.0.0.1::5060 5 31604 receiver child=0 sock=1 @ 195.37.77.101::5060 6 31605 receiver child=1 sock=1 @ 195.37.77.101::5060 7 31606 receiver child=2 sock=1 @ 195.37.77.101::5060 8 31610 receiver child=3 sock=1 @ 195.37.77.101::5060 9 31611 fifo server 10 31627 timer 3.1.5. Is Server Alive It is essential for solid operation to know continuously that server is alive. We've been using two tools for this purpose. sipsak does a great job of "pinging" a server, which may be used for alerting on unresponsive servers. monit is a server watching utility which alerts when a server dies. 3.1.6. Dealing with DNS SIP standard leverages DNS. Administrators of ser should be aware of impact of DNS on server's operation. Server's attempt to resolve an unresolvable address may block a server process in terms of seconds. To be safer that the server doesn't stop responding due to being blocked by DNS resolving, we recommend the following practices: * Start a sufficient number of children processes. If one is blocked, the other children will keep serving. * Use DNS caching. For example, in Linux, there is an nscd daemon available for this purpose. * Process transactions statefully if memory allows. That helps to absorb retransmissions without having to resolve DNS for each of them. * In your script expressions compare IP addresses without enclosing them in quotes. Enclosing IP addresses in quotes may cause additional reverse DNS lookups. Example 3-4. IP Address Comparison # this expression takes no DNS lookup if (src_ip==192.168.2.15) { .... # whereas this does if (src_ip=="192.168.2.15") { ... 3.1.7. Logging ser by default logs to syslog facility. It is very useful to watch log messages for abnormal behaviour. Log messages, subject to syslog configuration may be stored at different files, or even at remote systems. A typical location of the log file is /var/log/messages. Note One can also use other syslogd implementation. metalog ( http://metalog.sourceforge.net/ ) features regular expression matching that enables to filter and group log messages. For the purpose of debugging configuration scripts, one may want to redirect log messages to console not to pollute syslog files. To do so configure ser in the following way: * Attach ser to console by setting fork=no. * Set explicitely at which address ser should be listening, e.g., listen=192.168.2.16. * Redirect log messages to standard error by setting log_stderror=yes * Set appropriately high log level. (Be sure that you redirected logging to standard output. Flooding system logs with many detailed messages would make the logs difficult to read and use.) You can set the global logging threshold value with the option debug=nr, where the higher nr the more detailed output. If you wish to set log level only for some script events, include the desired log level as the first parameter of the log action in your script. The messages will be then printed if log's level is lower than the global threshold, i.e., the lower the more noisy output you get. Example 3-5. Logging Script # # $Id: logging.cfg,v 1.1 2003/02/27 20:29:25 jiri Exp $ # # logging example # # ------------------ module loading ---------------------------------- fork=no listen=192.168.2.16 log_stderror=yes debug=3 # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log(1, "REGISTER received\n"); } else { log(1, "non-REGISTER received\n"); }; if (uri=~"sip:.*[@:]iptel.org") { log(1, "request for iptel.org received\n"); } else { log(1, "request for other domain received\n"); }; } The following SIP message causes then logging output as shown bellow. REGISTER sip:192.168.2.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.33:5060 From: sip:113311@192.168.2.16 To: sip:113311@192.168.2.16 Call-ID: 00036bb9-0fd305e2-7daec266-212e5ec9@192.168.2.33 Date: Thu, 27 Feb 2003 15:10:52 GMT CSeq: 101 REGISTER User-Agent: CSCO/4 Contact: sip:113311@192.168.2.33:5060 Content-Length: 0 Expires: 600 [jiri@cat sip_router]$ ./ser -f examples/logging.cfg Listening on 192.168.2.16 [192.168.2.16]::5060 Aliases: cat.iptel.org:5060 cat:5060 WARNING: no fork mode 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(17379) REGISTER received 0(17379) request for other domain received 3.1.8. Labeling Outbound Requests Without knowing, which pieces of script code a relayed request visited, trouble-shooting would be difficult. Scripts typically apply different processing to different routes such as to IP phones and PSTN gateways. We thus recommend to label outgoing requests with a label describing the type of processing applied to the request. Attaching "routing-history" hints to relayed requests is as easy as using the append_hf action exported by textops module. The following example shows how different labels are attached to requests to which different routing logic was applied. Example 3-6. "Routing-history" labels # is the request for our domain? # if so, process it using UsrLoc and label it so. if (uri=~[@:\.]domain.foo") { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; # user found -- forward to him and label the request append_hf("P-hint: USRLOC\r\n"); } else { # it is an outbound request to some other domain -- # indicate it in the routing-history label append_hf("P-hint: OUTBOUND\r\n"); }; t_relay(); This is how such a labeled requests looks like. The last header field includes a label indicating the script processed the request as outbound. # U 2002/09/26 02:03:09.807288 195.37.77.101:5060 -> 203.122.14.122:5060 SUBSCRIBE sip:rajesh@203.122.14.122 SIP/2.0. Max-Forwards: 10. Via: SIP/2.0/UDP 195.37.77.101;branch=53.b44e9693.0. Via: SIP/2.0/UDP 203.122.14.115:16819. From: sip:rajeshacl@iptel.org;tag=5c7cecb3-cfa2-491d-a0eb-72195d4054c4. To: sip:rajesh@203.122.14.122. Call-ID: bd6c45b7-2777-4e7a-b1ae-11c9ac2c6a58@203.122.14.115. CSeq: 2 SUBSCRIBE. Contact: sip:203.122.14.115:16819. User-Agent: Windows RTC/1.0. Proxy-Authorization: Digest username="rajeshacl", realm="iptel.org", al gorithm="MD5", uri="sip:rajesh@203.122.14.122", nonce="3d924fe900000000 fd6227db9e565b73c465225d94b2a938", response="a855233f61d409a791f077cbe1 84d3e3". Expires: 1800. Content-Length: 0. P-hint: OUTBOUND. _________________________________________________________ 3.2. HOWTOs This section is a "cookbook" for dealing with common tasks, such as user management or controlling access to PSTN gateways. _________________________________________________________ 3.2.1. User Management There are two tasks related to management of SIP users: maintaining user accounts and maintaining user contacts. Both these jobs can be done using the serctl command-line tool. Also, the complimentary web interface, serweb, can be used for this purpose as well. If user authentication is turned on, which is a highly advisable practice, user account must be created before a user can log in. To create a new user account, call the serctl add utility with username, password and email as parameters. It is important that the environment SIP_DOMAIN is set to your realm and matches realm values used in your script. The realm value is used for calculation of credentials stored in subscriber database, which are bound permanently to this value. [jiri@cat gen_ha1]$ export SIP_DOMAIN=foo.bar [jiri@cat gen_ha1]$ serctl add newuser secret newuser@foo.bar MySql Password: new user added serctl can also change user's password or remove existing accounts from system permanently. [jiri@cat gen_ha1]$ serctl passwd newuser newpassword MySql Password: password change succeeded [jiri@cat gen_ha1]$ serctl rm newuser MySql Password: user removed User contacts are typically automatically uploaded by SIP phones to server during registration process and administrators do not need to worry about them. However, users may wish to append permanent contacts to PSTN gateways or to locations in other administrative domains. To manipulate the contacts in such cases, use serctl ul tool. Note that this is the only correct way to update contacts -- direct changes to back-end MySql database do not affect server's memory. Also note, that if persistence is turned off (usrloc "db_mode" parameter set to "0"), all contacts are gone on server reboot. Make sure that persistence is enabled if you add permanent contacts. To add a new permanent contact for a user, call serctl ul add . To delete all user's contacts, call serctl ul rm . serctl ul show prints all current user's contacts. [jiri@cat gen_ha1]$ serctl ul add newuser sip:666@gateway.foo.bar sip:666@gateway.foo.bar 200 Added to table ('newuser','sip:666@gateway.foo.bar') to 'location' [jiri@cat gen_ha1]$ serctl ul show newuser ;q=1.00;expires=1073741812 [jiri@cat gen_ha1]$ serctl ul rm newuser 200 user (location, newuser) deleted [jiri@cat gen_ha1]$ serctl ul show newuser 404 Username newuser in table location not found _________________________________________________________ 3.2.2. User Aliases Frequently, it is desirable for a user to have multiple addresses in a domain. For example, a user with username "john.doe" wants to be reachable at a shorter address "john" or at a nummerical address "12335", so that PSTN callers with digits-only key-pad can reach him too. With ser, you can maintain a special user-location table and translate existing aliases to canonical usernames using the lookup action from usrloc module. The following script fragment demonstrates use of lookup for this purpose. Example 3-7. Configuration of Use of Aliases if (!uri==myself) { # request not for our domain... route(1); # go somewhere else, where outbound requests are processed break; }; # the request is for our domain -- process registrations first if (method=="REGISTER") { route(3); break; }; # look now, if there is an alias in the "aliases" table; don't care # about return value: whether there is some or not, move ahead then lookup("aliases"); # there may be aliases which translate to other domain and for which # local processing is not appropriate; check again, if after the # alias translation, the request is still for us if (!uri==myself) { route(1); break; }; # continue with processing for our domain... ... The table with aliases is updated using the serctl tool. serctl alias add adds a new alias, serctl alias show prints an existing alias, and serctl alias rm removes it. [jiri@cat sip_router]$ serctl alias add 1234 sip:john.doe@foo.bar sip:john.doe@foo.bar 200 Added to table ('1234','sip:john.doe@foo.bar') to 'aliases' [jiri@cat sip_router]$ serctl alias add john sip:john.doe@foo.bar sip:john.doe@foo.bar 200 Added to table ('john','sip:john.doe@foo.bar') to 'aliases' [jiri@cat sip_router]$ serctl alias show john ;q=1.00;expires=1073741811 [jiri@cat sip_router]$ serctl alias rm john 200 user (aliases, john) deleted Note that persitence needs to be turned on in usrloc module. All changes to aliases will be otherwise lost on server reboot. To enable persistence, set the db_mode usrloc parameter to a non-zero value. # ....load module ... loadmodule "modules/usrloc/usrloc.so" # ... turn on persistence -- all changes to user tables are immediately # flushed to mysql modparam("usrloc", "db_mode", 1) # the SQL address: modparam("usrloc", "db_url","sql://ser:secret@dbhost/ser") _________________________________________________________ 3.2.3. Access Control (PSTN Gateway) It is sometimes important to exercise some sort of access control. A typical use case is when ser is used to guard a PSTN gateway. If a gateway was not well guarded, unauthorized users would be able to use it to terminate calls in PSTN, and cause high charges to its operator. There are few issues you need to understand when configuring ser for this purpose. First, if a gateway is built or configured to accept calls from anywhere, callers may easily bypass your access control server and communicate with the gateway directly. You then need to enforce at transport layer that signaling is only accepted if coming via ser and deny SIP packets coming from other hosts and port numbers. Your network must be configured not to allow forged IP addresses. Also, you need to turn on record-routing to assure that all session requests will travel via ser. Otherwise, caller's devices would send subsequent SIP requests directly to your gateway, which would fail because of transport filtering. Authorization (i.e., the process of determining who may call where) is facilitated in ser using group membership concept. Scripts make decisions on whether a caller is authorized to make a call to a specific destination based on user's membership in a group. For example a policy may be set up to allow calls to international destinations only to users, who are members of an "int" group. Before user's group membership is checked, his identity must be verified first. Without cryptographic verification of user's identity, it would be impossible to assert that a caller really is who he claims to be. The following script demonstrates, how to configure ser as an access control server for a PSTN gateway. The script verifies user identity using digest authentication, checks user's privileges, and forces all requests to visit the server. Example 3-8. Script for Gateway Access Control # # $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ # # example: ser configured as PSTN gateway guard; PSTN gateway is locate d # at 192.168.0.10 # # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/acc/acc.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/mysql/mysql.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" loadmodule "modules/group/group.so" loadmodule "modules/uri/uri.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit u s */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, dro p it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served her e"); } else { forward(uri:host, uri:port); }; break; }; # account completed transactions via syslog setflag(1); # free call destinations ... no authentication needed if ( is_user_in("Request-URI", "free-pstn") /* free destinatio ns */ | uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip:98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==192.168.0.10) { # our gateway doesn't support digest authentication; # verify that a request is coming from it by source # address log("gateway-originated request"); } else { # in all other cases, we need to check the request agai nst # access control lists; first of all, verify request # originator's identity if (!proxy_authorize( "gateway" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "gateway" /* realm */, "0" /* no qop */ ); break; }; # authorize only for INVITEs -- RR/Contact may result i n weird # things showing up in d-uri that would break our logic ; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single z ero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local") ) { sl_send_reply("403", "No permis sion for local calls"); break; }; # the same for long-distance (destinations begi n with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permi ssion for LD "); break; }; # the same for international calls (three zeros ) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "Internati onal permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go t o GW! rewritehostport("192.168.0.10:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } Use the serctl tool to maintain group membership. serctl acl grant makes a user member of a group, serctl acl show shows groups of which a user is member, and serctl acl revoke [] revokes user's membership in one or all groups. [jiri@cat sip_router]$ serctl acl grant john int MySql Password: +------+-----+---------------------+ | user | grp | last_modified | +------+-----+---------------------+ | john | int | 2002-12-08 02:09:20 | +------+-----+---------------------+ _________________________________________________________ 3.2.4. Accounting In some scenarios, like termination of calls in PSTN, SIP administrators may wish to keep track of placed calls. ser can be configured to report on completed transactions. Reports are sent by default to syslog facility. Support for RADIUS and mysql accounting exists as well. Note that ser is no way call-stateful. It reports on completed transactions, i.e., after a successful call set up is reported, it drops any call-related state. When a call is terminated, transactional state for BYE request is created and forgotten again after the transaction completes. This is a feature and not a bug -- keeping only transactional state allows for significantly higher scalability. It is then up to the accounting application to correlate call initiation and termination events. To enable call accounting, tm and acc modules need to be loaded, requests need to be processed statefuly and labeled for accounting. That means, if you want a transaction to be reported, the initial request must have taken the path "setflag(X), t_relay" in ser script. X must have the value configured in acc_flag configuration option. Also note, that by default only transactions that initiate a SIP dialog (typically INVITE) visit a proxy server. Subsequent transactions are exhanged directly between end-devices, do not visit proxy server and cannot be reported. To be able to report on subsequent transactions, you need to force them visit proxy server by turning record-routing on. Example 3-9. Configuration with Enabled Accounting # # $Id: acc.cfg,v 1.3 2003/06/03 03:18:12 jiri Exp $ # # example: accounting calls to nummerical destinations # # ------------------ module loading ---------------------------------- loadmodule "modules/tm/tm.so" loadmodule "modules/acc/acc.so" loadmodule "modules/sl/sl.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/rr/rr.so" # ----------------- setting module-specific parameters --------------- # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # Process record-routing if (loose_route()) { t_relay(); break; }; # labeled all transaction for accounting setflag(1); # record-route INVITES to make sure BYEs will visit our server too if (method=="INVITE") record_route(); # forward the request statefuly now; (we need *stateful* forwar ding, # because the stateful mode correlates requests with replies an d # drops retranmissions; otherwise, we would have to report on # every single message received) if (!t_relay()) { sl_reply_error(); break; }; } ______________________________ ___________________________ 3.2.5. Reliability It is essential to guarantee continuous service operation even under erroneous conditions, such as host or network failure. The major issue in such situations is transfer of operation to a backup infrastructure and making clients use it. The SIP standard's use of DNS SRV records has been explicitly constructed to handle with server failures. There may be multiple servers responsible for a domain and referred to by DNS. If it is impossible to communicate with a primary server, a client can proceed to another one. Backup servers may be located in a different geographic area to minimize risk caused by areal operational disasters: lack of power, flooding, earthquake, etc. Note Unless there are redundant DNS servers, fail-over capability cannot be guaranteed. Unfortunately, at the moment of writing this documentation (end of December 2002) only very few SIP products actually implement the DNS fail-over mechanism. Unless networks with SIP devices supporting this mechanism are built, alternative mechanisms must be used to force clients to use backup servers. Such a mechanism is disconnecting primary server and replacing it with a backup server locally. It unfortunately precludes geographic dispersion and requires network multihoming to avoid dependency on single IP access. Another method is to update DNS when failure of the primary server is detected. The primary drawback of this method is its latency: it may take long time until all clients learn to use the new server. The easier part of the redundancy story is replication of ser data. ser relies on replication capabilities of its back-end database. This works with one exception: user location database. User location database is a frequently accessed table, which is thus cached in server's memory to improve performance. Back-end replication does not affect in-memory tables, unless server reboots. To facilitate replication of user location database, server's SIP replication feature must be enabled in parallel with back-end replication. The design idea of replication of user location database is easy: Replicate any successful REGISTER requests to a peer server. To assure that digest credentials can be properly verified, both servers need to use the same digest generation secret and maintain synchronized time. A known limitation of this method is it does not replicate user contacts entered in another way, for example using web interface through FIFO server. The following script example shows configuration of a server that replicates all REGISTERs. Example 3-10. Script for Replication of User Contacts # # $Id: replicate.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ # # demo script showing how to set-up usrloc replication # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break;