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user-experience.xml
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user-experience.xml
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<?xml version="1.0" encoding="UTF-8"?>
<chapter xmlns="http://docbook.org/ns/docbook"
xmlns:xlink="http://www.w3.org/1999/xlink" version="5.0"
xml:id="userexperience">
<title>User Experience</title>
<para>Any successful IT project needs to focus on the needs of the user.
This is particularly important for communications technology.</para>
<sect1>
<title>First time setup and provisioning</title>
<para>The most successful RTC projects all have a convenient means of
user provisioning.</para>
<para>In an office environment, this may mean that the softphone or
desk phone is automatically configured for the user by a system
administrator. <xref linkend="useragents"/> discusses the
provisioning facilities of some types of phone.</para>
<para>For a generic SIP service, an ISP or a corporate/campus
environment with a bring-your-own (BYO) device policy, the ideal
setup process should only require the user to enter their SIP/XMPP
address and their password and the softphone/device should discover
all other parameters using DNS NAPTR and DNS SRV queries.
For this to be effective, softphones and devices aiming to be used
in this way need to ensure they have suitable codec and encryption
settings enabled by default so the user won't have to tweak them
as recommended in <xref linkend="optimal-connectivity"/>.</para>
</sect1>
<sect1 xml:id="userexperience-dialing">
<title>Dialing</title>
<para>At first glance, dialing a telephone may appear to be a
trivial task. Some attention to detail is required to maximize
user convenience.</para>
<sect2>
<title>Usernames or phone numbers?</title>
<para>A user may want to call or chat to various people. For some
people, they may only have the email/SIP/XMPP address and for
other people they may only have a phone number.</para>
<para>It is important to select phones that work
intuitively for either type of input, an address of the
form <literal>[email protected]</literal> or a phone number.
It is also important to ensure that the server processes, such
as SIP proxies, are correctly configured to route calls to
arbitrary Internet addresses. The repro SIP proxy and most
XMPP servers work this way by default.</para>
<para>Some phones can even help convert from a phone number to
a SIP or XMPP address, see <xref linkend="enum"/>.</para>
<para>If the user has an entry in their address book with both
a phone number and an email/SIP/XMPP address, then it is important
that the phone gives the user a helpful way to choose which one
to dial without asking too many questions about whether the user
wants to use SIP or XMPP. This process can be optimized if the
phone uses presence (a buddy list) to work out which contact
mechanisms are unreachable.</para>
<para>All of this can be made easier by using named accounts
(like email addresses) instead of extension numbers within the
server configuration, this is explained in more detail in
<xref linkend="user-authentication-usernames"/>.</para>
</sect2>
<sect2>
<title>Dial plans</title>
<para>Users find it easier to dial numbers in a local format
(without a country code). Many personal address books and
company databases store phone numbers in a local format. This can
lead to confusion in situations where somebody tries to use a
number in a different country, for example, if they take their
mobile/cell phone to another country and try to dial a number
stored in the address book.</para>
<para>A dial plan should be designed to convert phone numbers to
the international format (E.164), even if a user dials the number
in a local format. This makes it easier to send some calls to
carriers in different countries and it makes life easier for
organizations that expand into multiple countries. This means the
user can dial in the local or international format but the phone
system will work either way.</para>
</sect2>
<sect2>
<title>Dialing Internet addresses</title>
<para>In many cases, an email address may also be a SIP address or
XMPP address. Ideally, when a user tries to dial a contact from
their phone, the software should identify whether the contact is
reachable over SIP or XMPP and use that route for the call if
appropriate.</para>
</sect2>
</sect1>
</chapter>