applying the ring voltage.
In this example of the distributed call control model, R1 made
a local decision to send the call setup message to R2 based on
the call routing table of R1. R2 again made a local decision
(using the R2 call routing table) that the called device could
be reached on a certain physical port. - Supervise
the call: During the call, R1 and R2 both monitor the
quality of the call. In the distributed call control model, if
one of the gateways detects that the quality is no longer
acceptable, the gateway locally terminates the call.
Supervision of the call occurs during the second stage of the
call, call maintenance.
- Terminate the call: If
the caller that is connected to R1 finishes the call, R1
informs R2 of the termination. In the distributed call control
model, the gateway initiates the third stage of a call, call
teardown. The gateway terminates the call and releases the
resources that were used by the call.
Figure shows
how each gateway uses distributed call control to make its own
autonomous decisions and not depend on the availability of
another (centralized) device to provide call routing services.
Because each gateway has its own intelligence, there is no
single point of failure. However, each gateway needs to have a
local call routing table, which needs manual configuration.
This need makes administration of the distributed call control
model less scalable. For larger deployments using the
distributed call control model, managers add special devices
for centralized number lookup. Gateways and endpoints use H.323
gatekeepers or SIP network servers to find numbers that are not
known locally. In such a deployment, there is no need for all
(or even any) numbers to be stored at the gateways or
endpoints; numbers are stored only at the centralized devices.
Content 2.1 Introducing VoIP
Networks 2.1.7 Centralized Call Control
Figure shows an environment where a single component in the
network, a call agent, handles call control. Such centralized
call control applies when the voice-capable device does not
support call control on its own but uses a call agent. In this
example, both voice gateways have the MGCP protocol enabled.
With MGCP call control enabled, the gateways use the call agent
to perform these functions: - 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) informs its call
agent of the request.
- The call agent tells R1 to play
a dial tone and receive the digits that the user dials.
- R1 passes each received digit (one by one) to the call
agent.
- The call agent looks up the called number in
the agent’s call routing table. According to the call routing
table, the requested telephone number is reached using the
second gateway (R2). This call agent also controls R2 and,
therefore, knows which telephone numbers R2 can reach.
Therefore, the call agent knows which port at R2 the call has
to be routed to.
- The call agent now sends a message to
R2 requesting that the call pass to a certain port (the port
that connects to the destination telephone number).
Both call routing decisions, which gateway to use after
receiving the call at R1 and how to pass the call on at that
next gateway (R2), are made by the call agent. This is an
example of the centralized call control model, where all call
routing intelligence (required for the first stage of a call,
the call setup) is at the call agent. The call agent then
instructs the gateways on how to handle the call. So only the
call agent has a call routing table.
- Supervise the call: During the call, R1 and R2 both
monitor the quality of the call. In the centralized call
control model, if one of the gateways detects that the quality
is no longer adequate, it will pass that information to the
call agent. The call agent then terminates the call.
Supervision of the call occurs during the second stage of a
call, call maintenance.
- Terminate the call: If
the caller that is connected to R1 finishes the call, R1
informs the call agent of the termination. The call agent
notifies both gateways to terminate the VoIP call and release
the resources that were used by the call. In the centralized
call control model, the call agent initiates the third stage of
a call, call teardown.
As illustrated in the
example, with centralized call control, the gateways do not
make any local decisions. Instead, they inform the call agent
about events (such as incoming or dropped calls). Only the call
agent makes call routing decisions, and the gateways depend on
the availability of their call agent. Availability of the call
agent is critical, because the call agent is a single point of
failure. However, only the call agent needs to have a call
routing table. This makes administration of the centralized
call control model more scalable. Centralized call control
allows an external device (call agent) to handle signaling and
call processing, leaving the gateway to translate audio signals
into voice packets after call setup. After the call is set up,
the voice path runs directly between the two gateways and does
not involve the call agent. The difference between distributed
and centralized call control applies only to signaling and
never to the media exchange, which always goes on directly
between the two gateways.
Content 2.2
Digitizing and Packetizing Voice 2.2.1 Basic
Voice Encoding: Converting Analog Signals to Digital
Signals VoIP systems rely on digital signal processors
(DSPs). DSPs convert analog voice signals into digital format
and vice versa. They also provide functions such as voice
compression, transcoding (changing between different formats of
digitized voice), and conferencing. DSPs are hardware
components often located on voice modules inside gateways.
Sampling is the technique that is used to digitize
analog information. For example, producers digitize music for
CDs by sampling live sound at frequent intervals and then
digitizing each sample. Sampling is the reduction of a
continuous signal to a discrete signal. In converting analog
music sound waves (a continuous-time signal), DSPs build a
sequence of samples (a discrete-time signal) from which the
analog signal can be rebuilt for playing back on a CD player.
DSPs have a similar role in digitizing voice signals in
voice-enabled routers. Figure illustrates how voice-enabled
routers convert analog voice signals to digital format for
encapsulation in IP packets and transport over IP
networks.
In the example, a call is being made from an
analog telephone (Phone1), that is connected to a router (R1),
to an analog telephone (Phone2) that is connected to another
router (R2). The two routers connect to an IP network. The user
at Phone1 speaks into the microphone of the telephone, and the
telephone sends an analog signal to the FXS port of router R1.
Router R1 converts the received analog signal to a digital
signal and encapsulates the bits into IP packets. The IP
network carries the IP packets to router R2. DSPs on the voice
interface cards of the voice-enabled routers perform the
analog-to-digital conversion. Figure summarizes the steps:
Step 1 Sampling: The DSP periodically samples the
analog signal. The output of the sampling is a pulse amplitude
modulation (PAM) signal measured in volts. Step 2
Quantization: The DSP matches the PAM signal to a
segmented digital scale. This scale measures the amplitude
(height or voltage) of the PAM signal. Step 3
Compression: The DSP compresses voice samples to reduce
bandwidth requirements.
Content 2.2 Digitizing
and Packetizing Voice 2.2.2 Basic Voice
Encoding: Converting Digital Signals to Analog Signals When
a router receives voice input in digital format, it has to
convert it back to analog signals before sending it out to