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Indice de la Documentación iptel.org SIP Express Router v0.11.0 -- Admin's Guide

iptel.org SIP Express Router v0.11.0 -- Admin's Guide

Jiri Kuthan

Jan Janak

Yacine Rebahi

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 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 addresses:

    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 Foundation,
   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


-------------------------------------------------------------------------

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		     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 requests,
to keep the transport-translating SER in path. Also, if a destination
transport is not known, stateful forwarding is recommended -- use of
stateless forwarding for TCP2UDP would result in loss of reliability.


core
----
- reply_route has been renamed to failure_route -- the old name caused
  too much confusion
- forward_tcp and forward_udp can force SER to forward via specific
  transport protocol

acc module:
-----------
- radius and sql support integrated in this module; you need to
  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_request"
  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, group
  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 authentication
- group module contains group membership checking functions
- group_radius contains radius group membeship checking functions
- is_in_group has been renamed to is_user_in and places to groups
  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 saving
  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 AOR
  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 just
  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_relay
  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" /* table 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 URIs
            #  - 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
<sip:jiri@212.202.172.134>;q=0.00;expires=456
<sip:7271@gateway.foo.bar>;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 config), 
# 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/exectest.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;echo>/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 now 
	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_route[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