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<TH align=middle colSpan=3>3.3. Scenario 2: Alternatives to legacy
PBX systems</TH></TR>
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<TD align=left width="20%"><A accessKey=p
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch03s02.html">Prev</A> </TD>
<TH align=middle width="60%">Chapter 3. IP Telephony
Scenarios</TH>
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<DIV>
<H2 class=title style="CLEAR: both"><A id=d0e2177>3.3. Scenario 2:
Alternatives to legacy PBX systems</H2></DIV></DIV>
<DIV></DIV>
<P>Traditionally, institutions and companies are equipped with a PBX on each one
of its sites. Telephones are wired to the PBX, which supplies them with power.
The PBX handles all intelligence and routes calls to the PSTN over trunks (E1,
T1, J1, ISDN30 etc). </P>
<DIV class=figure><A id=scen_pbx>
<P class=title><B>Figure 3.4. Legacy PBX which trunks to the
PSTN</B></P>
<DIV class=mediaobject align=center><IMG
alt="Legacy PBX which trunks to the PSTN" src="ch03s03.files/scen_pbx.png"
align=middle></DIV></DIV>
<P>One of the most economically feasible deployments of IP Telephony currently
is in the area of installing voice over IP as a replacement of inter-building
PSTN connections within one company, or even the complete replacement of the PBX
phone system itself, along with its terminals.</P>
<P>This chapter first describes the scenarios in which IP phones can be deployed
in a peer-to-peer fashion without additional control entities in the network.
This case is only covered briefly because its practical use is limited.</P>
<P>Then a more common scenario will be described, where IP Telephony is
introduced in the existing telephony infrastructure. The legacy PBX is still
functional in this scenario, and voice calls can not only carried over regular
PSTN trunks, but also over IP backbones.</P>
<P>The last scenario describes the total replacement of a PBX infrastructure by
IP Telephony equipment.</P>
<DIV class=sect2 lang=en>
<DIV class=titlepage>
<DIV>
<DIV>
<H3 class=title><A id=d0e2196>3.3.1. Scenario 2a: IP-Phones without a PBX
system</H3></DIV></DIV>
<DIV></DIV>
<P>The simplest case of IP Telephony is making a point-to-point call between IP
Phones. To place a call, the calling party needs to know the IP address of the
called party, or its DNS entry.</P>
<DIV class=figure><A id=scen_phone-phone>
<P class=title><B>Figure 3.5. IP-Phone to IP-Phone without PBX</B></P>
<DIV class=mediaobject align=center><IMG alt="IP-Phone to IP-Phone without PBX"
src="ch03s03.files/scen_phone-phone.png" align=middle></DIV></DIV>
<P>For mission-critical cases, such as a commercial company or an institutional
phone system, this is an awkward method. Moreover, it is not possible to make a
call to a regular telephone within the institution or to the PSTN, because no
VoIP-to-PSTN gateway is available. Also, common features like central address
books, call forwarding services, etc. are harder to integrate with the phone
terminal. If the destination is unreachable, nothing useful can be done with the
call, like redirecting it to a voicemail service, etc. This setup is therefore
only recommended for testing purposes.</P>
<P>Call setup is very simple, when using either H.323, or SIP or any variations
of these protocols. Since the calling party directly enters the IP address of
the destination, call initiation signaling is sent directly to that IP address.
If the terminal is functional, it will process the signaling and the called
party will be prompted to pick up the phone. When that happens, the call is
setup, a codec is negotiated and the voice stream will start, until signaling
that terminates the call is exchanged. </P></DIV>
<DIV class=sect2 lang=en>
<DIV class=titlepage>
<DIV>
<DIV>
<H3 class=title><A id=d0e2211>3.3.2. Scenario 2b: Integration of VoIP with
legacy PBX systems</H3></DIV></DIV>
<DIV></DIV>
<P>This scenario allows the coexistence and intercommunication of the
institutional conventional telephony network (conventional phones connected to
PBX) and the local IP telephony network. The scenario is suitable when the local
IP telephony network is constructed gradually in an institution that already has
a conventional telephony network. In a later stage, the conventional telephony
network and the PBX can be totally replaced by the IP telephony network, thus
converging to <A
title="3.3.3. Scenario 2c: Full replacement of legacy PBX systems"
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch03s03.html#fullreplacement">Scenario
2c</A>. </P>
<P>For example, in order to provide for smooth transition, it might be
worthwhile to buy a gateway with two ISDN PRI interfaces (or just with one
interface and borrow the second interface for the transition period). One
interface is connected to the PSTN and the second one to PBX. During the
transition period Gateway performs also call routing between PSTN and the old
PBX and vice versa providing a smooth transition in the meanwhile.</P>
<P>In this chapter we give an overview of options for interconnecting a PBX to a
Voice Gateway (VoGW). These options also apply to <A
title="3.2. Scenario 1: Long-distance least cost routing"
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch03s02.html">Scenario
1</A>. More technical details for individual interfaces are given in <A
title="Chapter 4. Setting up basic services"
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04.html">Chapter
4</A>.</P>
<DIV class=figure><A id=scen_hybrid>
<P class=title><B>Figure 3.6. Integration of IP-Telephony with legacy
PBX system</B></P>
<DIV class=mediaobject align=center><IMG
alt="Integration of IP-Telephony with legacy PBX system"
src="ch03s03.files/scen_hybrid.png" align=middle></DIV></DIV>
<P>The choice of a particular interface between a PBX and a VoGW depends on
required functionality, availability of interconnection ports on both sides and
also on cost constraints. Interfaces can be divided into analog and digital. The
former include a 2-wire U-interface with subscriber loop and various types of
E&M interfaces. The latter include an E1/CAS trunk with MFC-R2 signaling and
ISDN with DSS1 or QSIG signaling. Giving technical details about the trunks and
interfaces mentioned above is outside the scope of this Chapter, refer to <A
title="Chapter 4. Setting up basic services"
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04.html">Chapter
4</A> for further details. On the other hand, technical people who want to
understand this kind of scenario may benefit from a discussion of the advantages
and shortcomings of individual interfaces, which are summarized in the following
list:</P>
<DIV class=itemizedlist>
<UL type=disc>
<LI><SPAN class=emphasis><EM>Subscriber loop</EM></SPAN> - Suitable when
conventional phones should be connected directly to VoGW (Voice GateWay) via
an FXS interface - an FXS interface connects directly to a standard telephone
and supplies ring, voltage, and dial tone, but can also be used for PBX
interconnection. A disadvantage is that when calling inward towards the PBX,
an extension number can be dialed only as DTMF (Dual-Tone Multifrequency)
suffix, after a call is established and is already accounted for. This type of
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