Introducing VoIP Networks 2.1.3
Legacy Analog Interfaces in VoIP Networks A VoIP network
that includes legacy equipment, such as analog telephones,
needs gateways to convert analog signals into digital format
and encapsulate them into IP packets. A network built with
IP-enabled devices needs no conversion. Gateways use different
types of interfaces to connect to analog devices, such as
telephones, fax machines, or PBX or PSTN switches. Analog
interfaces that are used at the gateways include these three
types: - Foreign Exchange Station (FXS): FXS is
a telephone interface that provides battery power, sends a dial
tone, and generates ringing voltage. A standard telephone plugs
into such an interface to receive telephone service. A
telephone exchange is an example of an FXS. In VoIP
implementations, the FXS interface connects to analog end
systems; the analog end system uses the Foreign Exchange Office
(FXO) interface on the end system side. The router FXS
interface behaves like a PSTN or a PBX by serving telephones,
answering machines, or fax machines with line power, ring
voltage, and dial tones. If a PBX uses an FXO interface, it can
also connect to a router FXS interface. In this case, the PBX
acts like a telephone.
- Foreign Exchange Office
(FXO): An FXO is a telephone interface that connects to the
PSTN. FXO generates the on-hook and off-hook indicators that
are used to signal a loop closure at the FXO’s end of the
circuit. In VoIP implementations, the FXO interface connects to
the PSTN or a PBX; both the PSTN and PBX use the FXS interface
on their side. The router FXO interface behaves like a
telephone, obtaining line power, ring voltage, and dial tones
from the other side of the interface. As mentioned, a PBX can
also use an FXO interface toward the router (which will then
use an FXS interface) if the PBX takes the role of the
telephone.
- Ear and Mouth (E&M): The
E&M interface provides signaling for analog trunks. Analog
trunks interconnect two PBX-style devices, such as any
combination of a gateway (acting as a PBX), a PBX, and a PSTN
switch. E&M is often defined as “ear and mouth,” but it
derives from the term “earth and magnet.” “Earth” represents
the electrical ground, and “magnet” represents the
electromagnet used to generate tones.
In Figure ,
the gateway serves a telephone and a fax machine using two FXS
interfaces. For these two devices, the router acts like a PBX
or a PSTN switch. The router connects to the PSTN using an FXO
interface. For this connection, the router acts like a
telephone toward the PSTN. Another FXO interface connects to a
PBX (PBX-1). Again, the router acts like an end system toward
the PBX, and hence uses the same port type as the telephone and
fax that connect to PBX-1 use. A second PBX (PBX-2) connects to
the router FXS interface. For this connection, the PBX behaves
like a telephone toward the router, and the router acts like a
PSTN switch. Finally, the router connects to another PBX
(PBX-3), this time using an E&M interface. On this trunk
connection, both the router and PBX-3 act as a PBX.
Content 2.1 Introducing VoIP Networks
2.1.4 Digital Interfaces Gateways can use
digital interfaces to connect to voice equipment. From a
hardware perspective, there are BRIs and T1 and E1 interfaces
available. All of the interfaces use TDM to support multiple
logical channels. T1 and E1 interfaces can use either channel
associated signaling (CAS) or common channel signaling (CCS),
while a BRI always uses CCS. ISDN PRIs use T1 or E1 CCS. Figure
illustrates a router that serves an ISDN telephone using a BRI
voice interface. In addition, the router has two T1 or E1
lines: one to a PBX and one to the PSTN. Depending on the
interface type (T1 or E1) and the signaling method (CAS or
CCS), a maximum of 23, 24, or 30 voice channels are available
on these trunks. The PSTN and the PBX also serve ISDN
telephones through BRI connections. The table in Figure shows
the number of voice channels, the signaling bandwidth, total
bandwidth, and framing overhead for all available interface and
signaling options.
Content 2.1
Introducing VoIP Networks 2.1.5 Stages for
Completing a VoIP Telephone Call Although different
protocols deal with call control in different ways, all
protocols provide a common set of services. There are three
components of basic call control: - Call setup:
Call setup checks the call-routing configuration to determine
the destination of a call. The configuration specifies the
bandwidth requirements for the call. Knowing the bandwidth
requirements, CAC determines whether sufficient bandwidth is
available to support the call. If bandwidth is available, call
setup generates a setup message and sends the message to the
destination. If bandwidth is not available, call setup presents
the call initiator with a busy signal. Different call control
protocols, such as H.323, MGCP, and Session Initiation Protocol
(SIP), define different sets of messages that devices exchange
during setup. However, all of the messages consult the same
basic information:
- The IP addresses of the two devices
that will exchange VoIP
- The User Datagram Protocol
(UDP) port numbers that will be used for the Real-Time
Transport Protocol (RTP) streams which carry the voice
traffic
- The format (for example, the compression
algorithm) used for the digitized voice
-
Call maintenance: Call maintenance tracks packet count,
packet loss, jitter, and delay during the call. Information
passes to the voice-enabled devices to determine whether
connection quality is good or has deteriorated to the point
where the gateway will drop the call.
- Call
teardown: Call teardown notifies voice-enabled devices to
make resources available for other calls and make them
available for other calls when either side terminates a
call.
Note
Later lessons in this module
explain the protocols mentioned here. Return to this section to
place these protocols in perspective.
Content
2.1 Introducing VoIP Networks
2.1.6 Distributed Call Control There are two types
of call control: distributed and centralized. In the past, all
voice networks used a centralized architecture in which dumb
endpoints (telephones) were controlled by centralized switches.
Although this model worked well for basic telephony services,
it mandated a trade-off between simplified management and
endpoint and service innovation. One of the benefits of VoIP
technology is that it allows networks to use either a
centralized or a distributed architecture. This flexibility
allows companies to build networks characterized by both
simplified management and endpoint innovation, depending on the
protocol used. Figure illustrates the distributed model in
which multiple components in the network handle call control.
Configuring the voice-capable device to support call control
directly enables distributed call control using protocols such
as H.323 or SIP. With distributed call control, the devices
perform the call setup, call maintenance, and call teardown
without telephone company involvement: - Process
dialed digits and route the call (sequence shown in
Figure ):
- After detecting a service request (the
telephone is taken off the hook), the first gateway (R1) plays
a dial tone.
- Next, R1 collects the digits that the
caller dials.
- R1 then looks up the called number in
the R1 local call routing table. According to the call routing
table, the called number uses the second gateway (R2).
- R1 now enters the first stage of a call, call setup, by
sending the appropriate message to R2.
- R2 receives
the call setup message from R1.
- R2 then looks up the
called number in its local call routing table. According to the
call routing table, the called number uses a local voice
port.
- R2 sends the call to the local voice port by