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.

1.5. 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

  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


    

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.

# 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.

# 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.

# 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 (()).

# 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")
# ...

# 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();
        };
}			

if (uri=~"^sip:(.+@)?(192\.168\.0\.10|(sip\.)?foo\.bar)([:;\?].*)?$")
      log(1, "yes, it is a request for our domain");
      break;
 };
			

# 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.

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.

# 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.

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.

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).

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
		    

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).

[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.

# 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.

# 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 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.

# ------------------ 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.

  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.

#
# $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.

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();
		};
   	};
}


		    

			
			#
# $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.

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");
	};
}


		    

			
			#
# $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();
}

		    

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="3d9385170000000043acbf6ba9c9741790e0c57adee73812", 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.

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", algorithm="MD5", uri="sip:rajesh@203.122.14.122", nonce="3d924fe900000000fd6227db9e565b73c465225d94b2a938", response="a855233f61d409a791f077cbe184d3e3".
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.

[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
			    

[jiri@cat gen_ha1]$ serctl passwd newuser newpassword
MySql Password: 
password change succeeded
[jiri@cat gen_ha1]$ serctl rm newuser                
MySql Password: 
user removed
			    

[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
<sip:666@gateway.foo.bar>;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
			    

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...
...
  
				

[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                    
<sip:john.doe@foo.bar>;q=1.00;expires=1073741811
[jiri@cat sip_router]$ serctl alias rm john  
200 user (aliases, john) deleted				
			    

# ....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")
			    

#
# $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $
#
# example: ser configured as PSTN gateway guard; PSTN gateway is located
# 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 us */
	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, drop 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 here");
		} 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 destinations */
			|  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 against
		# 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 in 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 zero)? 
			# (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 permission for local calls"); 
					break;
				};
			# the same for long-distance (destinations begin with two zeros")
			} else if (uri=~"sip:00[1-9][0-9]+@.*") {
				if (!is_user_in("credentials", "ld")) {
					sl_send_reply("403", " no permission 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", "International 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 to GW!

	rewritehostport("192.168.0.10:5060");

	# forward the request now
	if (!t_relay()) {
		sl_reply_error(); 
		break; 
	};

}

				

[jiri@cat sip_router]$ serctl acl grant john int
MySql Password: 
+------+-----+---------------------+
| user | grp | last_modified       |
+------+-----+---------------------+
| john | int | 2002-12-08 02:09:20 |
+------+-----+---------------------+
			    

#
# $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* forwarding,
	# because the stateful mode correlates requests with replies and
	# 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.

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;
	};

	# 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") {

			# verify credentials
			if (!www_authorize("foo.bar", "subscriber")) {
				www_challenge("foo.bar", "0");
				break;
			};

			# if ok, update contacts and ...
			save("location");
			# ... if this REGISTER is not a replica from our
			# peer server, replicate to the peer server
			if (!src_ip==backup.foo.bar) {
				t_replicate("backup.foo.bar", "5060");
			};
			break;
		};
		# do whatever else appropriate for your domain
		log("non-REGISTER\n");
	};
}

				    
				

loadmodule("modules/tm/tm.so");
loadmodule("modules/acc/acc.so");
....
# if a call is labeled using setflag(3) and is missed, it will
# be reported
...
modparam("acc", "log_missed_flag", 3 );
if (!lookup("location")) {
     # call invitations to off-line users are reported using the
     # acc_request action; to avoid duplicate reports on request
     # retransmissions, request is processed statefuly (t_newtran,
     # t_reply)
     if ((method=="INVITE" || method=="ACK") && t_newtran() ) {
          t_reply("404", "Not Found");
	  acc_request("404 Not Found");
          break;
     };
     # all other requests to off-line users are simply replied
     # statelessly and no reports are issued
    sl_send_reply("404", "Not Found");
    break;
} else {
     # user on-line; report on failed transactions; mark the
     # transaction for reporting using the same number as 
     # configured above; if the call is really missed, a report
     # will be issued
     setflag(3);
     # forward to user's current destination
     t_relay();
     break;
};
			    

---

3.2.9. NAT Traversal

NATs are worst things that ever happened to SIP. These devices are very popular because they help to conserve IP address space and save money charged for IP addresses. Unfortunately, they translate addresses in a way which is not compatible with SIP. SIP advertises receiver addresses in its payload. The advertised addresses are invalid out of NATted networks. As a result, SIP communication does not work accross NATs without extra effort.

There are few methods that may be deployed to traverse NATs. How proper their use is depends on the deployment scenario. Unfortunatelly, all the methods have some limitations and there is no straight-forward solution addressing all scenarios. Note that none of these methods takes explicit support in ser.

The first issue is whether SIP users are in control of their NATs. If not (NATs are either operated by ISP or they are sealed to prevent users setting them up), the only method is use of a STUN-enabled phone. STUN is a very simple protocol used to fool NAT in such a way, they permit SIP sessions. Currently, we are aware of one softphone (kphone) and one hardphone (snom) with STUN support, other vendors are working on STUN support too. Unfortunately, STUN gives no NAT traversal guarantee -- there are types of NATs, so called symmetric NATs, over which STUN fails to work.

There is actually yet another method to address SIP-unaware, user-uncontrolled NATs. It is based on a proxy server, which relays all signaling and media and mangles packets to make them more NAT-friendly. The very serious problem with this method is it does not scale.

If users are in control of their own NAT, as typically residential users are, they can still use STUN. However, they may use other alternatives too. One of them is to replace their NAT with a SIP-aware NAT. Such NATs have built-in SIP awareness, that patches problems caused by address translations. Prices of such devices are getting low and there are available implementations (Intertex, Cisco/PIX). No special support in phones is needed.

Other emerging option is UPnP. UPnP is a protocol that allows phones to negotiate with NAT boxes. You need UPnP support in both, NAT and phones. As UPnP NATs are quite affordable, costs are not an obstacle. Currently, we are aware of one SIP phone (SNOM) with UPnP support.

Geeks not wishing to upgrade their firewall to a SIP-aware or UPnP-enabled one may try to configure static address translation. That takes phones with configuration ability to use fixed port numbers and advertise outside address in signaling. Cisco phones have this capability, for example. The NAT devices need to be configured to translate outside port ranges to the ranges configured in phones.

---

3.2.11. Authentication Policy: Prevention of Unauthorized Domain Name Use in From and More

Malicous users can claim a name of domain, to which they do not administratively belong, in From header field. This behaviour cannot be generally prevented. The reason is that requests with such a faked header field do not need to visit servers of the domain in question. However, if they do so, it is desirable to assure that users claiming membership in a domain are actually associated with it. Otherwise the faked requests would be relayed and appear as coming from the domain, which would increase credibility of the faked address and decrease credibility of the proxy server.

Preventing unathorized domain name use in relayed requests is not difficult. One needs to authenticate each request with name of the served domain in From header field. To do so, one can search for such a header field using search action (textops module) and force authentication if the search succeeds.

A straight-forward solution might be to authenticate ALL requests. However, that only works in closed networks in which all users have an account in the server domain. In open networks, it is desirable to permit incoming calls from callers from other domains without any authentication. For example, a company may wish to accept calls from unknown callers who are new prospective customers.

# does the user claim our domain "foo.bar" in From?
if (search("^(f|From):.*foo.bar")) {
        # if so, verify credential
	if (!proxy_authorize("foo.bar", "subscriber")) { 
              # don't proceed if credentials broken; challenge
	      proxy_challenge("foo.bar", "0");
	      break;
        };
};
		    

In general, the authentication policy may be very rich. You may not forget each request deserves its own security and you need to decide whether it shall be authenticated or not. As mentioned above, in closed networks, you may want to authenticate absolutely every request. That however prohibits traffic from users from other domains. A pseudo-example of a reasonable policy is attached: it looks whether a request is registration, it claims to originate from our domain in From header field, or is a local request to another domain.

# (example provided by Michael Graff on [serusers] mailing list
if (to me):
    if register
          www_authorize or fail if not a valid register
          done
    if claiming to be "From" one of the domains I accept registrations for
          proxy_authorize
          done
    if not to me (I'm relaying for a local phone to an external address)
          proxy_authorize
          done
		    

You also may want to apply additional restriction to how digest username relates to usernames claimed in From and To header fields. For example, the check_to action enforces the digest id to be equal to username in To header fields. That is good in preventing someone with valid credentials to register as someone else (e.g., sending a REGISTER with valid credentials of "joe" and To belonging to "alice"). Similarly, check_from is used to enforce username in from to equal to digest id.

There may be a need for a more complex relationship between From/To username and digest id. For example, providers with an established user/password database may wish to keep using it, whereas permitting users to claim some telephone numbers in From. To address such needs generally, there needs to be a 1:N mapping between digest id and all usernames that are acceptable for it. This is being addressed in a newly contributed module "domain", which also addresses more generally issues of domain matching for multidomain scenarios.

Other operational aspect affecting the authentication policy is guarding PSTN gateways (see Section 3.2.3). There may be destinations that are given away for free whereas other destinations may require access control using group membership, to which authentication is a prerequisity.

# $Id: ccdiversion.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $

# ------------------ module loading ----------------------------------

loadmodule "modules/textops/textops.so"
loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"

# ----------------- setting module-specific parameters ---------------

route{
	# if we do not get a positive reply, continue at reply_route[2]
	t_on_failure("2");
	# forward the request to all destinations in destination set now 
	t_relay();
}

failure_route[2] {
	# request failed (no reply, busy, whatever) ... forward it again
	# to pbx's voicemail at phone number 6000 via Cisco gateway at
	# 192.168.10.10; append proprietary CC-Diversion header field with
	# original request uri in it, so that the gateway and voicemail
	# know, whom the request was originally intended for
	append_branch("sip:6000@192.168.10.10");
	append_urihf("CC-Diversion: ", "\r\n");
	t_relay();
}


			

3.3. Troubleshooting

This section gathers practices how to deal with errors known to occur frequently. To understand how to watch SIP messages, server logs, and in genereal how to troubleshoot, read also Section 3.1.

    if (len_gt( max_len )) {
        sl_send_reply("513", "Message too big");
        break;
    };

    # DON'T FORGET THE FOLLOWING LINES
    if (loose_route()) {
        t_relay();
        break;
    };

4.1. Using exec Module

The easiest way is to couple ser with external applications via the exec module. This module allows execution of logic and URI manipulation by external applications on request receipt. While very simple, many useful services can be implemented this way. External applications can be written in any programming language and do not be aware of SIP at all. ser interacts with the application via standard input/output and environment variables.

For example, an external shell script may send an email whenever a request for a user arrives.

# send email if a request for user "jiri" arrives
if (uri=~"^sip:jiri@") {
     exec_msg("echo 'body: call arrived'|mail -s 'call for you' jiri");
}
		

In this example, the exec_msg action starts an external shell. It passes a received SIP request to shell's input. In the shell, mail command is called to send a notification by e-mail. The script however features several simplifications:

1. The email notification does not tell who was calling.

2. The logic is not general: it only supports one well-known user (jiri).

3. The logic is stateless. It will be executed on every retransmission.

4. It is a script fragment not explaining the context. This particular example may be for example used to report on missed calls.

#
# $Id: exec_s2.cfg,v 1.1 2003/01/05 20:37:51 jiri Exp $
#
# send a notificiation if a request for user jiri arrives
# ------------------ module loading ----------------------------------

loadmodule "modules/exec/exec.so"
loadmodule "modules/sl/sl.so"

# -------------------------  request routing logic -------------------

# main routing logic

route{
	# send email if a request for jiri arrives
	if (uri=~"^sip:jiri@") {
		exec_msg("(egrep -i '^(From|f):'; 
			echo 'request received')|
			mail -s 'request for you' jiri");
	};

}


		

INVITE sip:jiri@iptel.org SIP/2.0
Via: SIP/2.0/UDP 195.37.77.100:5040
Max-Forwards: 10
From: "alice" <sip:alice@iptel.org>;tag=76ff7a07-c091-4192-84a0-d56e91fe104f
To: <sip:jiri@iptel.org>
Call-ID: d10815e0-bf17-4afa-8412-d9130a793d96@213.20.128.35
CSeq: 2 INVITE
Contact: <sip:123.20.128.35:9315>
Content-Type: application/sdp
Content-Length: 451

--- SDP payload snipped ---

		    

Date: Thu, 12 Dec 2002 14:25:02 +0100
From: root <root@cat.iptel.org>
To: jiri@cat.iptel.org
Subject: request for you

From: "alice" <sip:alice@iptel.org>;tag=76ff7a07-c091-4192-84a0-d56e91fe104f
request received
		    

# send email if a request for "jiri" arrives
if (uri=~"^sip:jiri@") {
     exec_msg("echo request received from $SIP_HF_FROM | mail -s 'request for you' jiri");
};
		

SIP_HF_MAX_FORWARDS=10
SIP_HF_VIA=SIP/2.0/UDP 195.37.77.100:5040
SIP_HF_CSEQ=2 INVITE
SIP_HF_FROM="alice" <sip:alice@iptel.org>;tag=76ff7a07-c091-4192-84a0-d56e91fe104f
SIP_ORUI=sip:jiri@iptel.org
SIP_HF_CONTENT_LENGTH=451
SIP_TID=3b6b8295db0835815847b1f35f3b29b8
SIP_DID=
SIP_RURI=iptel.org
SIP_HF_TO=<sip:jiri@iptel.org>
SIP_OUSER=jiri
SIP_HF_CALLID=d10815e0-bf17-4afa-8412-d9130a793d96@213.20.128.35
SIP_SRCIP=195.37.77.100
SIP_HF_CONTENT_TYPE=application/sdp
SIP_HF_CONTACT=<sip:123.20.128.35:9315>
		    

#
# $Id: exec_s3.cfg,v 1.2 2003/01/25 15:04:43 janakj Exp $
#
# email notification to email address from mysql database
#

# ------------------ module loading ----------------------------------

loadmodule "modules/exec/exec.so"
loadmodule "modules/sl/sl.so"

# send email if a request arrives
route[0] {
     if (!exec_msg('
	QUERY="select email_address from subscriber 
		where user=\"$SIP_OUSER\"";
	EMAIL=`mysql  -Bsuser -pheslo -e "$QUERY" ser`;
	if [ -z "$EMAIL" ] ; then exit 1; fi ;
	echo "SIP request received from $SIP_HF_FROM for $SIP_OUSER" |
	mail -s "request for you" $EMAIL ')) {
		# exec returned error ... user does not exist
		sl_send_reply("404", "User does not exist");
	} else {
		sl_send_reply("600", "No messages for this user");
	};
		
}

		    

Example 4-4. Adding Stateful Processing

The previously improved example still features a shortcoming. When a message retransmission arrives due to a nework mistake such as lost reply, the email notification is executed again and again. That happens because the script is stateless, i.e., no track of current transactions is kept. The script does not know whether a request is a new or a retransmitted one. Transaction management may be introduced by use of tm module as described in Section 2.7.2. In the script, t_newtran is first called to absorb requests retransmission -- if they occur, script does not continue. Then, as in the previous example, an exec module action is called. Eventually, a reply is sent statefully.

Note carefuly: it is important that the stateful reply processing (t_reply) is used as opposed to using stateless replies (sl_send_reply). Otherwise, the outgoing reply would not affect transactional context and would not be resent on receipt of a request retransmission.

#
# $Id: exec_s4.cfg,v 1.2 2003/01/25 15:04:43 janakj Exp $
#
# email notification to email address from mysql database
#

fork=no

# ------------------ module loading ----------------------------------

loadmodule "modules/exec/exec.so"
loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"

# send email if a request arrives; process statefully
# to avoid multiple execution on request retransmissions
route[0] {
	# stop script processing if transaction exists
	if ( !t_newtran()) {
		sl_reply_error();
		break;
	};

     	if (!exec_msg('
		QUERY="select email_address from subscriber 
			where user=\"$SIP_OUSER\"";
		EMAIL=`mysql  -Bsuser -pheslo -e "$QUERY" ser`;
		if [ -z "$EMAIL" ] ; then exit 1; fi ;
		echo "SIP request received from $SIP_HF_FROM for $SIP_OUSER" |
		mail -s "request for you" $EMAIL ')) 
	{
		# exec returned error ... user does not exist
		# send a stateful reply
		t_reply("404", "User does not exist");
	} else {
		t_reply("600", "No messages for this user");
	};
		
}

		    

#
# $Id: exec_s5.cfg,v 1.2 2003/01/25 15:04:43 janakj Exp $
#
# simple quick-start config script
#
fork=no
log_stderror=yes
# ----------- global configuration parameters ------------------------

loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"
loadmodule "modules/usrloc/usrloc.so"
loadmodule "modules/registrar/registrar.so"
loadmodule "modules/exec/exec.so"

# ----------------- setting module-specific parameters ---------------

route{
	# uri for my domain ?
	if (uri==myself) {

		if (method=="REGISTER") {
			save("location");
			break;
		};

		# native SIP destinations are handled using our USRLOC DB
		if (!lookup("location")) {
			# proceed to email notification
			if (method=="INVITE") route(1)
			else sl_send_reply("404", "Not Found");
			break;
		};
	};
	# user found, forward to his current uri now
	if (!t_relay()) {
		sl_reply_error();
	};
}

/* handling of missed calls */
route[1] {
	# don't continue if it is a retransmission
	if ( !t_newtran()) {
		sl_reply_error();
		break;
	};
	# external script: lookup user, if user exists, send 
	# an email notification to him
     	if (!exec_msg('
		QUERY="select email_address from subscriber 
			where user=\"$SIP_OUSER\"";
		EMAIL=`mysql  -Bsuser -pheslo -e "$QUERY" ser`;
		if [ -z "$EMAIL" ] ; then exit 1; fi ;
		echo "SIP request received from $SIP_HF_FROM for $SIP_OUSER" |
		mail -s "request for you" $EMAIL ')) 
	{
		# exec returned error ... user does not exist
		# send a stateful reply
		t_reply("404", "User does not exist");
	} else {
		t_reply("600", "No messages for this user");
	};
	break;
}

		    

#
# $Id: exec_s5b.cfg,v 1.5 2003/06/20 23:20:35 jiri Exp $
#
# simple quick-start config script
#
fork=no
log_stderror=yes
# ----------- global configuration parameters ------------------------

loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"
loadmodule "modules/usrloc/usrloc.so"
loadmodule "modules/registrar/registrar.so"
loadmodule "modules/exec/exec.so"

# ----------------- setting module-specific parameters ---------------

route{
	# uri for my domain ?
	if (uri==myself) {

		if (method=="REGISTER") {
			save("location");
			break;
		};

		# native SIP destinations are handled using our USRLOC DB
		if (!lookup("location")) {
			# proceed to email notification
			if (method=="INVITE") route(1)
			else sl_send_reply("404", "Not Found");
			break;
		};
	};
	# user found, forward to his current uri now; if any 
	# forwarding error occurs (e.g., busy or cancelled recevied
	# from downstream), proceed to reply_route[1]
	t_on_failure("1");
	if (!t_relay()) {
		sl_reply_error();
	};
}

/* handling of missed calls */
route[1] {
	# don't continue if it is a retransmission
	if ( !t_newtran()) {
		sl_reply_error();
		break;
	};
	# external script: lookup user, if user exists, send 
	# an email notification to him
     	if (!exec_msg('
		QUERY="select email_address from subscriber 
			where username=\"$SIP_OUSER\"";
		EMAIL=`mysql  -Bsuser -pheslo -e "$QUERY" ser`;
		if [ -z "$EMAIL" ] ; then exit 1; fi ;
		echo "SIP request received from $SIP_HF_FROM for $SIP_OUSER" |
		mail -s "request for you" $EMAIL ')) 
	{
		# exec returned error ... user does not exist
		# send a stateful reply
		t_reply("404", "User does not exist");
	} else {
		t_reply("600", "No messages for this user");
	};
	break;
}

failure_route[1] {
	# just call exec, that's it
   	exec_msg('
		QUERY="select email_address from subscriber 
			where username=\"$SIP_OUSER\"";
		EMAIL=`mysql  -Bsuser -pheslo -e "$QUERY" ser`;
		if [ -z "$EMAIL" ] ; then exit 1; fi ;
		echo "SIP request received from $SIP_HF_FROM for $SIP_OUSER" |
		mail -s "request for you" $EMAIL ') ;
	t_relay();
}

		    

4.2. Application FIFO Server

Application FIFO server is a very powerful method to program SIP services. The most valuable benefit is it works with SIP-unaware applications written in any programming language. Textual nature of the FIFO interface allows for easy integration with a lot of existing programs. Today, ser's complementary web-interface, serweb, written in PHP, leverages the FIFO interface when displaying and changing user location records stored in server's memory. It uses this interface to send instant messages too, without any knowledge of underlying SIP stack. Another application relying on the FIFO interface is serctl, ser management utility. The command-line utility can browse server's in-memory user-location database, display running processes and operational statistics.

The way the FIFO server works is similar to how /proc filesystem works on some operating systems. It provides a human-readable way to access ser's internals. Applications dump their requests into the FIFO server and receive a status report when request processing completes. ser exports a lot of its functionality located in both the core and external modules through the FIFO server.

FIFO requests are formed easily. They begin with a command enclosed in colons and followed by name of file or pipe (relative to /tmp/ path), to which a reply should be printed. The first request line may be followed by additional lines with command-specific parameters. For example, the t_uac_dlg FIFO command for initiating a transaction allows to pass additional header fields and message body to a newly created transaction. Each request is terminated by an empty line. Whole requests must be sent by applications atomically in a single batch to avoid mixing with requests from other applications. Requests are sent to pipe at which ser listens (filename configured by the fifo config file option).

An easy way to use the FIFO interface is via the serctl command-line tool. When called along with "fifo", FIFO command name, and optional parameters, the tool generates a FIFO request and prints request result. The following example shows use of this tool with the uptime and which commands. uptime returns server's running time, which returns list of available FIFO commands. Note that only the built-in FIFO command set is displayed as no modules were loaded in this example.

[jiri@cat test]$ serctl fifo uptime
Now: Fri Dec  6 17:56:10 2002
Up Since: Fri Dec  6 17:56:07 2002
Up time: 3 [sec]

[jiri@cat test]$ serctl fifo which
ps
which
version
uptime
print
		    

:uptime:ser_receiver_1114
		    

[jiri@cat sip_router]$ serctl fifo foobar
500 command 'foobar' not available
		    

[jiri@fox ser]$ serctl fifo ul_show_contact location jiri
<sip:195.37.78.160:14519>;q=0.00;expires=1012
			

The user location example demonstrates an essential feature of the FIFO server: extensibility. It is able to export new commands implemented in new modules. Currently, usrloc module exports FIFO commands for maintaining in-memory user location database and tm module exports FIFO commands for management of SIP transactions. See the example in examples/web_im/send_im.php for how to initiate a SIP transaction (instant message) from a PHP script via the FIFO server. This example uses FIFO command t_uac_dlg. The command is followed by parameters: header fields and message body. The same FIFO command can be used from other environments to send instant messages too. The following example shows how to send instant messages from a shell script.

#!/bin/sh
#
# call this script to send an instant message; script parameters
# will be displayed in message body
#
# paremeters mean: message type, request-URI, outbound server is
# left blank ("."), required header fields From and To follow,
# then optional header fields terminated by dot and optional
# dot-terminated body

cat > /tmp/ser_fifo <<EOF
:t_uac_dlg:hh
NOTIFY
sip:receiver@127.0.0.1
.
From: sip:originator@foo.bar
To: sip:receiver@127.0.0.1
foo: bar_special_header
x: y
p_header: p_value
Contact: <sip:devnull@192.168.0.100:9>
Content-Type: text/plain; charset=UTF-8
.
Hello world!!!! $@
.
EOF
		    

/* construct and send a FIFO command; the command parameters $sip_address, 
   $expires are PHP variables originating from an HTML form
 */
	$fifo_cmd=":ul_add:".$config->reply_fifo_filename."\n".
		$config->ul_table."\n".			//table
		$user_id."\n".		//username
		$sip_address."\n".				//contact
		$expires."\n".					//expires
		$config->ul_priority."\n\n";		//priority
		$message=write2fifo($fifo_cmd, $errors, $status);

/* .......... snip .................. */

/* this is the stub function for communicating with FIFO server.
   it dumps a request to FIFO server, opens a reply FIFO and
   reads server's reply from it
*/
function write2fifo($fifo_cmd, &$errors, &$status){
	global $config;

	/* open fifo now */
	$fifo_handle=fopen( $config->fifo_server, "w" );
	if (!$fifo_handle) {
		$errors[]="sorry -- cannot open fifo"; return;
	}
	
	/* create fifo for replies */
	@system("mkfifo -m 666 ".$config->reply_fifo_path );

	/* add command separator */
	$fifo_cmd=$fifo_cmd."\n";
	
	/* write fifo command */
	if (fwrite( $fifo_handle, $fifo_cmd)==-1) {
	    @unlink($config->reply_fifo_path);
	    @fclose($fifo_handle);
		$errors[]="sorry -- fifo writing error"; return;
	}
	@fclose($fifo_handle);
	
	/* read output now */
	@$fp = fopen( $config->reply_fifo_path, "r");
	if (!$fp) {
	    @unlink($config->reply_fifo_path);
		$errors[]="sorry -- fifo reading error"; return;
	}

	$status=fgetS($fp,256);
	if (!$status) {
	    @unlink($config->reply_fifo_path);
		$errors[]="sorry -- fifo reading error"; return;
	}
	
	$rd=fread($fp,8192);
	@unlink($config->reply_fifo_path);
	
	return $rd;
}
		   
		

See Section 6.5 for a complete listing of FIFO commands available with current ser distribution.

        CTD                  Caller               Callee
            #1 INVITE
            ----------------->
                             ...
                             caller answers
            #2 200
            <-----------------
            #3 ACK
            ----------------->
            #4 REFER
            ----------------->
            #5 202
            <-----------------
            #6 BYE
            ----------------->
            #7 200
            <-----------------
                                  #8 INVITE
                                  ------------------>
                                  #9 180 ringing
                                  <------------------


#1 click-to-dial (CTD) is started and the "wrapper call" is initiated
INVITE caller
From: controller
To: caller
SDP: on hold

#2 calling user answes
200 OK
From: controller
To: caller

#3 CTD acknowledges
ACK caller
From controller
To: caller

#4 CTD initiates a transfer
REFER caller
From: controller
To: caller
Refer-To: callee
Refered-By: controller

#5 caller confirms delivery of REFER
202 Accepted
From: controller
To: caller

#6 CTD terminates the wrapper call -- it is no longer needed
BYE caller
From: controller
To: caller

#7 BYE is confirmed
200 Ok
From: controller
To: caller

#8 caller initates transaction solicited through REFER
INVITE callee
From: caller
To: callee
Referred-By: controller

#9 that's it -- it is now up to callee to answer the INVITE
180 ringing
From: caller
To: callee
			

[jiri@cat examples]$ ./ctd.sh 
destination unspecified -- taking default value sip:23@192.168.2.16
caller unspecified -- taking default value sip:113311@192.168.2.16
invitation succeeded
refer succeeded
bye succeeded
		    

5.1. serctl command-line tool

serctl is a command-line utility which allows to perform most of management tasks needed to operate ser: adding users, changing their passwords, watching server status, etc. Usage of utility is as follows:

usage: 
           * subscribers *
 serctl add <username> <password> <email> .. add a new subscriber (*)
 serctl passwd <username> <passwd> ......... change user's password (*)
 serctl rm <username> ...................... delete a user (*)
 serctl mail <username> .................... send an email to a user
 serctl alias show [<alias>] ............... show aliases
 serctl alias rm <alias> ................... remove an alias
 serctl alias add <alias> <uri> ............ add an aliases 

           * access control lists *
 serctl acl show [<username>] .............. show user membership
 serctl acl grant <username> <group> ....... grant user membership (*)
 serctl acl revoke <username> [<group>] .... grant user membership(s) (*)

           * usrloc *
 serctl ul show [<username>]................ show in-RAM online users
 serctl ul rm <username> ................... delete user's UsrLoc entries
 serctl ul add <username> <uri> ............ introduce a permanent UrLoc entry
 serctl showdb [<username>] ................ show online users flushed in DB

           * server health *
 serctl monitor ............................ show internal status
 serctl ps ................................. show runnig processes 
 serctl fifo ............................... send raw commands to FIFO

   Commands labeled with (*) will prompt for a MySQL password.
   If the variable PW is set, the password will not be prompted.

 

Prior to using the utility, you have to first set the environment variable SIP_DOMAIN to locally appropriate value (e.g., "foo.com"). It is needed for calculation of user credentials, which depend on SIP digest realm. (see also MSM Authentication Issue)

[cycle #: 2; if constant make sure server lives and fifo is on]
Server: Sip EXpress router(0.8.8 (i386/linux))
Now: Thu Sep 26 23:16:48 2002
Up Since: Thu Sep 26 12:35:27 2002
Up time: 38481 [sec]

Transaction Statistics
Current: 0 (0 waiting) Total: 606 (0 local)       
Replied localy: 34      
Completion status 6xx: 0, 5xx: 1, 4xx: 86, 3xx: 0,2xx: 519      

Stateless Server Statistics
200: 6218 202: 0 2xx: 0      
300: 0 301: 0 302: 0 3xx: 0      
400: 0 401: 7412 403: 2 404: 1258 407: 116 408: 0 483: 0 4xx: 25      500: 0 5xx: 0      
6xx: 0      
xxx: 0      
failures: 0      

UsrLoc Stats
Domain Registered Expired
'aliases' 9 0
'location' 29 17

		

5.2. Web User Provisioning -- serweb

To make provisioning of user accounts convenient, a web front-end to ser, serweb has been developed. serweb, a PHP-written web application, allows users to apply for new ser accounts, and maintain these. Users can manipulate their contacts, keep a phone-book with SIP addresses, change password, send instant SIP messages, and more. Administrators can manipulate any accounts and grant or revoke user privileges.

serweb is freely available from berlios site at http://developer.berlios.de/cvs/?group_id=500. Installation takes unpacking tarball to a safe destination at web server (better not in the HTML tree) and configuring config.php accordingly to local conditions.

Running serweb can be seen at iptel.org's SIP site. Just create and use a SIP account at http://www.iptel.org/user/

5.3. Voicemail

#
# $Id: ser.cfg,v 1.11 2003/07/03 12:17:56 rco Exp $
#
# iptel's voicemail config script
#

# ----------- global configuration parameters ------------------------

debug=7		        # debug level (cmd line: -dddddddddd)
fork=no

log_stderror=yes	# (cmd line: -E)
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 ----------------------------------

loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"
loadmodule "modules/maxfwd/maxfwd.so"
loadmodule "modules/rr/rr.so"
loadmodule "modules/textops/textops.so"
loadmodule "modules/vm/vm.so"
loadmodule "modules/dbtext/dbtext.so"
	    
# ----------------- setting module-specific parameters ---------------

modparam("voicemail", "db_url","/home/rco/cvs/ser/sip_router/modules/vm/db")

# -------------------------  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;
	};

	# 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
	loose_route();

	# Make MSN Messenger happy... 
	if (method=="REGISTER") {
		sl_send_reply("200","ok");
		break;
	};

	if (uri == myself) {

		# Voicemail specific configuration - begin

		if(method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER"){

			if(t_newtran()){

				t_reply("100","Trying -- just wait a minute !");

				if(method=="INVITE" || method=="REFER"){
					log("**************** vm start - begin ******************\n");
					if( uri =~ "conference" ){
						if(!vm("/tmp/am_fifo","conference")){
							log("could not contact conference server\n");
							t_reply("500","could not contact conference server");
						};
					}
					else if( uri =~ "echo" ){
						if(!vm("/tmp/am_fifo","echo")){
							log("could not contact echo\n");
							t_reply("500","could not contact echo");
						};
					}
					else {
						if(!vm("/tmp/am_fifo","voicemail")){
							log("could not contact voicemail\n");
							t_reply("500","could not contact voicemail");
						};
					};
					log("**************** vm start - end ******************\n");
					break;
				};
				
				if(method=="BYE"){
					log("**************** vm end/refer - begin ******************\n");
					if(!vm("/tmp/am_fifo","bye")){
						log("could not contact the media server\n");
						t_reply("500","could not contact the media server");
					};
					log("**************** vm end/refer - end ********************\n");
					break;
				};
			}
			else {
			     log("could not create new transaction\n");
			     sl_send_reply("500","could not create new transaction");
			};
		};

		# Voicemail specific configuration - end
	}
	else {
		if (!t_relay()) {
			sl_reply_error();
		};
	};
}











		

6.1. Core Options

Core options are located in beginning of configuration file and affect behaviour of the server.

• debug - Set log level, this is number between 0 and 9. Default is 0.

• fork - If set to yes, the server will spawn children. If set to no, the main process will be processing all messages. Default is yes.

  Disabling child spawning is useful mainly for debugging. When fork is turned off, some features are unavailable: there is no attendant process, no pid file is generated, and server listens only at one address. Make sure you are debugging the same interface at which ser listens. The easiest way to do so is to set the interface using listen option explicitly.

• log_stderror - If set to yes, the server will print its debugging information to standard error output. If set to no, syslog will be used. Default is no (printing to syslog).

• listen - list of all IP addresses or hostnames SER should listen on.

  This parameter may repeat several times, then SER will listen on all addresses. For example, the following command-line options (equivalent to "listen" config option) may be used: ser -l foo -l bar -p 5061 -l x -l y will listen on foo:5060, bar:5061 & x:5061 & y:5061

• alias - Add IP addresses or hostnames to list of name aliases. All requests with hostname matching an alias will satisfy the condition "uri==myself".

• dns - Uses dns to check if it is necessary to add a "received=" field to a via. Default is no.

• rev_dns - Same as dns but use reverse DNS. Default is no.

• port - Listens on the specified port (default 5060). It applies to the last address specified in listen and to all the following that do not have a corresponding "port" option.

• maxbuffer - Maximum receive buffer size which will not be exceeded by the auto-probing procedure even if the OS allows. Default value is MAX_RECV_BUFFER_SIZE, which is 256k.

• children - Specifies how many processes should be started for each transport protocol. Running multiple children allows a server to server multiple requests in parallel when request processing block (e.g., on DNS lookup). Note that ser typically spawns additional processes, such as timer process or FIFO server. If FIFO server is turned on, you can watch running processes using the serctl utility.

• check_via - Turn on or off Via host checking when forwarding replies. Default is no.

• syn_branch - Shall the server use stateful synonym branches? It is faster but not reboot-safe. Default is yes.

• memlog - Debugging level for final memory statistics report. Default is L_DBG -- memory statistics are dumped only if debug is set high.

• sip_warning - Should replies include extensive warnings? By default yes, it is good for trouble-shooting.

• fifo - FIFO special file pathname, for example "/tmp/ser_fifo". Default is no filename -- no FIFO server is started then. We recommend to set it so that accompanying applications such as serweb or serctl can communicate with ser.

• fifo_mode - Permissions of the FIFO special file.

• server_signature - Should locally-generated messages include server's signature? By default yes, it is good for trouble-shooting.

• reply_to_via - A hint to reply modules whether they should send reply to IP advertised in Via. Turned off by default, which means that replies are sent to IP address from which requests came from.

• user | uid - uid to be used by the server.

• group | gid - gid to be used by the server.

• mhomed -- enable calculation of outbound interface; useful on multihomed servers, ser .

• loadmodule - Specifies a module to be loaded (for example "/usr/lib/ser/modules/tm.so")

• modparam - Module parameter configuration. The commands takes three parameters:

  • module - Module in which the parameter resides.

  • parameter - Name of the parameter to be configured.

  • value - New value of the parameter.

6.2. Core Commands

Route Blocks and Process Control

• route[number]{...} - This marks a "route block" in configuration files. route blocks are basic building blocks of ser scripts. Each route block contains a sequence of SER actions enclosed in braces. Multiple route blocks can be included in a configuration file. When script execution begins on request receipt, route block number 0 is entered. Other route blocks serve as a kind of sub-routines and may be entered by calling the action route(n), where n is number of the block. The action break exits currently executed route block. It stops script execution for route block number 0 or returns to calling route block otherwise.

  Example 6-1. route

  route[0] {
          # call routing block number 2
  	route(2);
  }
  
  route[2] {
      forward("host.foo.bar", 5060);
  }

• failure_route is used to restart request processing when a negative reply for a previously relayed request is received. It is only used along with tm module, which stores the original requests and can return to their processing later. To activate processing of a failure_route block, call the TM action t_on_failure(route_number) before calling t_relay. When a negative reply comes back, the desired failure_route will be entered and processing of the original request may continue.

  The set of actions applicable from within failure_route blocks is limited. Permitted actions are URI-manipulation actions, logging and sending stateful replies using t_reply.

  Example 6-2. failure_route

  failure_route[1] {
      # for some reason, the original forwarding attempt
      # failed, try at another URI
      append_branch("sip:nonsense@iptel.org");
      # if this new attempt fails too, try another failure_route
      t_on_failure("2");
  	t_relay();
  }

• The action break exits currently executed route block. It stops script execution for route block number 0 or returns to calling route block otherwise.

  We recommend to use break after any request forwarding or replying. This practice helps to avoid erroneous scripts that continue execution and mistakenly send another reply or forward a request to another place, resulting in protocol confusion.

  Example: break;

• route(n) - call routing block route[n]{...}; when the routing block n finishes processing, control is passed back to current block and processing continues.

• if (condition) statement - Conditional statement.

  Example 6-3. Use of if

  if (method=="REGISTER) {
      log("register received\n");
  };

• if - else - If-Else Conditional statement.

  Example 6-4. Use of if-else

  if (method=="REGISTER) {
      log("register received\n");
  } else {
      log("non-register received\n");
  };

Flag Manipulation

• setflag - Set flag in the message.

  Example: setflag(1);

• resetflag - Reset flag in the message.

  Example: resetflag(1);

• isflagset - Test whether a particular flag is set.

  Example 6-5. isflagset

  if (isflagset(1)) {
      ....
  };

Manipulation of URI and Destination Set

• rewritehost | sethost | seth - Rewrite host part of the Request URI.

  Example: sethost("foo.bar.com");

• rewritehostport | sethostport | sethp - Rewrite host and port part of the Request URI.

  Example: sethostport("foo.bar.com:5060");

• rewriteuser | setuser | setu - Rewrite or set username part of the Request URI.

  Example: setuser("joe");

• rewriteuserpass | setuserpass | setup - Rewrite or set username and password part of the Request URI.

  Example: setuserpass("joe:mypass");

• rewriteport | setport | setp - Rewrite or set port of the Request URI.

  Example: setport("5060");

• rewriteuri | seturi - Rewrite or set the whole Request URI.

  Example: seturi("sip:joe@foo.bar.com:5060");

• revert_uri - Revert changes made to the Request URI and use original Request URI.

  Example: revert_uri();

• prefix - Add prefix to username in Request URI.

  Example: prefix("123");

• strip - Remove first n characters of username in Request URI.

  Example: strip(3);

• append_branch - Append a new destination to destination set of the message.

  Example 6-6. Use of append_branch

  # redirect to these two destinations: a@foo.bar and b@foo.bar
  # 1) rewrite the current URI
  rewriteuri("sip:a@foo.bar");
  # 2) append another entry to the destination ser
  append_branch("sip:b@foo.bar");
  # redirect now
  sl_send_reply("300", "redirection");
  			    

Message Forwarding

• forward(uri, port) - Forward the request to given destination statelessly. The uri and port parameters may take special values 'uri:host' and 'uri:port' respectively, in which case SER forwards to destination set in current URI. All other elements in a destination set are ignored by stateless forwarding.

  Example: forward("foo.bar.com"); # port defaults to 5060

• send - Send the message as is to a third party

  Example: send("foo.bar.com");

Logging

• log([level], message) - Log a message.

  Example: log(1, "This is a message with high log-level set to 1\n");

  Logging is very useful for troubleshooting or attracting administrator's attention to unusual situations. ser reports log messages to syslog facility unless it is configured to print them to stderr with the log_stderr configuration option. Log messages are only issued if their log level exceeds threshold set with the debug configuration option. If log level is omitted, messages are issued at log level 4.

Miscellaneous

• len_gt - If length of the message is greater than value given as parameter, the command will return 1 (indicating true). Otherwise -1 (indicating false) will be returned. It may take 'max_len' as parameter, in which case message size is limited to internal buffer size BUF_SIZE (3040 by default).

  Example 6-7. Use of len_gt

  # deny all requests larger in size than 1 kilobyte
  if (len_gt(1024)) {
      sl_send_reply("513", "Too big");
      break;
  };
  			

6.3. Command Line Parameters

Command-Line parameters may be overridden by configuration file options which take precedence over them.

• -h - Displays a short usage description, including all available options.

• -c - Performs loop checks and computes branches.

• -r - Uses dns to check if it is necessary to add a "received=" field to a via.

• -R - Same as -r but uses reverse dns.

• -v - Turns on via host checking when forwarding replies.

• -d - Turns on debugging, multiple -d increase debugging level.

• -D - Runs ser in the foreground (it doesn't fork into daemon mode).

• -E - Sends all the log messages to stderr.

• -V - Displays the version number.

• -f config-file - Reads the configuration from "config-file" (default ./ser.cfg).

• -l address - Listens on the specified address. Multiple -l mean listening on multiple addresses. The default behaviour is to listen on all the ipv4 interfaces.

• -p port - Listens on the specified port (default 5060). It applies to the last address specified with -l and to all the following that do not have a corresponding -p.

• -n processes-no - Specifies the number of children processes forked per interface (default 8).

• -b max_rcv_buf_size - Maximum receive buffer size which will not be exceeded by the auto-probing procedure even if the OS allows.

• -m shared_mem_size - Size of the shared memory which will be allocated (in Megabytes).

• -w working-dir - Specifies the working directory. In the very improbable event that will crash, the core file will be generated here.

• -t chroot-dir - Forces ser to chroot after reading the config file.

• -u uid - Changes the user id under which ser runs.

• -g gid - Changes the group id under which ser runs.

• -P pid-file - Creates a file containing the pid of the main ser process.

• -i fifo-path - Creates a fifo, useful for monitoring ser status.

6.4. Modules

Module description is currently located in READMEs of respective module directories. README-MODULES lists all available modules, including their maturity status. In the current ser distribution, there are the following modules:

• acc -- call accounting using syslog facility. RADIUS and mysql support can be compiled in. Depends on tm.

• auth, auth_db, auth_radius -- digest authentication. Depends on sl and mysql.

• domain -- checks URIs whether they belong in a list of served domains or not.

• ENUM -- E.164 phone number resolution using ENUM.

• exec -- execution of shell programs.

• group, group_radius -- checks membership of users in groups

• jabber -- gateway between SIMPLE and Jabber instant messaging. Depends on tm and mysql.

• maxfwd -- checking max-forwards header field.

• msilo -- message silo. Store for undelivered instant messages. Depends on tm and mysql.

• mysql -- mysql database back-end.

• nathelper -- facilitates NAT traversal for symmetric SIP phones such as ATA.

• pa -- presence agent

• registrar, usrloc -- User Location database. Works in in-memory mode or with mysql persistence support. Depends on sl, and on mysql if configured for use with mysql.

• rr -- Record Routing (strict and loose)

• sl -- stateless User Agent server.

• sms -- SIMPLE/SMS gateway. Depends on tm. Takes special hardware.

• textops -- textual database back-end.

• tm -- transaction manager (stateful processing).

• uri, uri_radius -- checks digest identity against header URIs or a database list

The most frequently used actions exported by modules are summarized in Table 6-1. For a full explanation of module actions, see documentation in respective module directories in source distribution of ser.

6.5. FIFO Commands Reference

This section lists currently supported FIFO commands. Some of them are built-in in ser core, whereas others are exported by modules. The most important exporters are now tm and usrloc module. tm FIFO commands allow users to initiate transactions without knowledge of underlying SIP stack. usrloc FIFO commands allow users to access in-memory user-location database. Note that that is the only way how to affect content of the data-base in real-time. Changes to MySql database do not affect in-memory table unless ser is restarted.

6.6. Used Database Tables

ser includes MySQL support to guarantee data persistence across server reboots and storage of users' web environment. The data stored in the database include user profiles, access control lists, user location, etc. Note that users are not supposed to alter the data directly, as it could introduce inconsistency between data on persistence storage and in server's memory. The following list enumerates used tables and explains their purpose.

• subscriber -- table of users. It includes user names and security credentials, as well as additional user information.

• reserved -- reserved user names. serweb does not permit creation of accounts with name on this list.

• phonebook -- user's personal phonebooks. Accessible via serweb.

• pending -- table of unconfirmed subscription requests. Used by serweb.

• missed_calls -- table of missed calls. Can be fed by acc modules if mysql support is turned on. Displayed by serweb.

• location -- user contacts. Typically updated through ser'r registrar functionality.

• grp -- group membership. Used by auth module to determine whether a user belongs to a group.

• event -- allows users to subscribe to additional services. Currently unused.

• aliases -- keeps track of alternative user names.

• active_sessions -- keeps track of currently active web sessions. For use by serweb.

• acc -- keeps track of accounted calls. Can be fed by acc module if mysql support is turned on. Displayed by serweb.

• config -- maintains attribute-value pairs for keeping various information. Currently not used.

• silo -- message store for instant messages which could not have been delivered.

• version -- keeps version number of each table definition.