FreeSWITCH 1.0.6: SIP and the User Directory

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(For more resources on Telephony, see here.)

Understanding the FreeSWITCH user directory

The FreeSWITCH user directory is based on a centralized XML document, comprised of one or more <domain> elements. Each <domain> can contain either <user> elements or <groups> elements. A <groups> element contains one or more <groups> elements, each of which contains one or more <user> elements. A small, simple example would look like the following:

<section name="directory">
<domain name="example.com">
<groups>
<group name="default">
<user id="1001">
<params>
<param name="password" value="1234"/>
</params>
</user>
</group>
</groups>
</domain>
</section>

Some more basic configurations may not have a need to organize the users in groups so it is possible to omit the <groups> element completely, and just insert several <user> elements into the top <domain> element.

The important thing is that each user@domain derived from this directory is available to all components in the system—it’s a single centralized directory for storing all of your user information. If you register as a user with a SIP phone or if you try to leave a voicemail message for a user, FreeSWITCH looks in the same place for user data. This is important because it limits duplication of data, and makes it more efficient than it would be if each component kept track of its users separately.

This system should work well for a small system with a few users in it, but what about a large system with thousands of users? What if a user wants to connect his existing database to FreeSWITCH to provide the user directory? Well, using mod_xml_curl (download here-ch:1,ch:3), we can create a web service that gets the request for the entries in the user directory, in the same way a web page sends the results of a form submission. In turn, that web service can query an existing database of users formatted any way possible, and construct the XML records in the format that FreeSWITCH registry expects. mod_xml_curl returns the data to the module requesting the lookup. This means that instant, seamless integration with your existing setup is possible; your data is still kept in its original, central location.

The user directory can be accessed by any subsystem within FreeSWITCH. This includes modules, scripts, and the FSAPI interface among others. In this article, we are going to learn how the Sofia SIP module employs the user directory to authenticate your softphone or hardware SIP phone. If you are a developer you may appreciate some nifty things you can do with your user directory, such as adding a <variables> element to either the <domain>, the <groups>, or the <user> element. In this element you can set many <variable> elements, allowing you to set channel variables that will apply to every call made by a particular authenticated user. This can come in very handy in the Dialplan because it allows you to make user-specific routing decisions. It is also possible to define IP address ranges using CIDR notation, which can be used to authenticate particular users based on what remote network address they connect from. This removes the need for a login and password, if your user always logs in from the same remote IP address.

The directory is implemented in pure XML. This is advantageous for several reasons, not the least of which is the “X” in XML: Extensible. Since XML is, by definition, extensible, the directory structure is also extensible. If we need to add a new element into the directory, we can do so simply by adding to the existing XML structure.

Authentication versus authorization
Authentication is the process of identifying a user. Authorization is the process of determining the level of access of a user. Authentication answers the question, “Is this person really who he says he is?” Authorization answers the question, “What is this person allowed to do here?” When you see expressions such as “IP Auth” and “Digest Auth”, remember that they are referring to the two primary ways of identifying (that is, authenticating) a user. IP authorization is based upon the user’s IP address. Digest authentication is based upon the user supplying a username and password. SIP (and FreeSWITCH) can use either method. Visit http://en.wikipedia.org/wiki/Digest_access_authentication for a discussion of how digest authentication works

Working with the FreeSWITCH user directory

The default configuration has one domain with a directory of 20 users. Users can be added or removed very easily. There is no set limit to how many users can be defined on the system. The list of users is collectively referred to as the directory. Users can belong to one or more groups. Finally, all the users belong to a single domain. By default, the domain is the IP address of the FreeSWITCH server.

In the following sections we will discuss these topics:

  • User features
  • Adding a user
  • Testing voicemail
  • Groups of users

User features

Let’s begin by looking at the XML file that defines a user. Locate the file conf/directory/default/1000.xml and open it in an editor. You should see a file like the following:

<include>
<user id="1000">
<params>
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow"
value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension
1000"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>

The XML structure of a user is simple. Within the <include> tags the user has the following:

  • The user element with the id attribute
  • The params element, wherein parameters are specified
  • The variables element, wherein channel variables are defined

Even before we know what much of the specifics mean, we can glean from this file that the user id is 1000 and that there is both a password and a vm-password. In this case, the password parameter refers to the SIP authorization password. The expression $${default_password} refers to the value contained in the global variable default_password which is defined in the conf/vars.xml file. If you surmised that vm-password means “voicemail password” then you are correct. This value refers to the digits that the user needs to dial when logging in to check his or her voicemail messages. The value of id is used both as the authorization username and the SIP username.

Additionally, there are a number of channel variables that are defined for this user. Most of these are directly related to the default Dialplan. The following table lists each variable and what it is used for:

Variable

Purpose

toll_allow

Specifies which types of calls this user can make

accountcode

Arbitrary value that shows up in CDR data

user_context

The Dialplan context that is used when this person makes a phone call

effective_caller_id_name

Caller ID name displayed on called party’s phone when calling another registered user

effective_caller_id_number

Caller ID number displayed on called party’s phone when calling another registered user

outbound_caller_id_name

Caller ID name sent to provider on outbound calls

outbound_caller_id_number

Caller ID number sent to provider on outbound calls

callgroup

Arbitrary value that can be used in Dialplan or CDR

In summary, a user in the default configuration has the following:

  • A username for SIP and for authorization
  • A voicemail password
  • A means of allowing/restricting dialling
  • A means of handling caller ID being sent out
  • Several arbitrary variables that can be used or ignored as needed

Let’s now add a new user to our directory.

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