Chương 2: Mạng VoIP

2.1. Tổng quan VoIP

2.2. Kỹ thuật VoIP
2.3. VoIP Gateway
2.4. Media Gateway Controller
2.5. Chuẩn ITU-T Rec. H.323
2.6. Giao thức SIP

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 2.1. Tổng quan VoIP

 What Is Voice Over IP (VoIP)?

-VoIP is a technology that enables routing of voice
communications through the Internet or any other
Internet protocol (IP)-based networks
-Voice is transmitted over a general-purpose packet-
switched network instead of dedicated traditional
circuit-switched voice transmission lines
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 2.1. Tổng quan VoIP

-The goals of VoIP implementation are to achieve

(a) significant savings in network maintenance and
operations costs and (b) rapid rollout of new
services.

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 2.1. Tổng quan VoIP

 VoIP: Why Implement It?

The success of VoIP can be attributed to the following key
reasons.
-Ease of deployment
-Simplification of transport networks
-Cost reduction
-Value-added services
-Anytime, anywhere communication
-Easy upgradation 4
 2.1. Tổng quan VoIP

-VoIP could enable a service provider to transport

voice for “free” over the Internet as transport of
packets over IP network is free. However, as VoIP
operates over IP which is the “best-effort” protocol,
it requires certain QoS guarantees.

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 2.1. Tổng quan VoIP

 Evolution of VoIP
-With the rapid growth of Internet and deregulation
of the telecommunications industry, infrastructure
convergence in the form of building voice applications
on top of data networks led to the birth of VoIP
-There are two fundamental technologies that are
necessary for the existence of VoIP, namely telephone
and Internet 6
2.1. Tổng quan VoIP

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 2.1. Tổng quan VoIP

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 2.1. Tổng quan VoIP

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 2.2. Kỹ thuật VoIP

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 2.2. Kỹ thuật VoIP
1. IP-enabled workstation:
 -The end users that are involved in VoIP communication must have
IP-enabled devices that are compatible with VoIP protocols and
allow routing through IP-based networks.
 -The workstation can be a soft phone installed in a computer with
access to an IP network.
 -IP-enabled mobile and fixed telephones can also be used to
implement VoIP technology.

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 2.2. Kỹ thuật VoIP
2. VoIP server:
 -The VoIP server is the centralized node that initiates, manages, and
terminates communication between the caller and the callee.
 The VoIP server must implement the call signaling protocols (SIP,
H.323, etc.) and ensure proper routing of IP packets to their
destination.
 Call admission control is one of the primary functions of the server.
 It can also be used for QoS provisioning mechanisms.

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 2.2. Kỹ thuật VoIP
3. Gateway:
 One way to increase the interoperability of VoIP is by
implementing it in diverse networks having different characteristics
which is made possible using gateways.
 Gateways ensure proper coordination in between these networks
and further allow VoIP users to communicate to PSTN-based
telephones.
 Moreover, firewalls can be implemented in gateways to achieve
secured communication with appropriate packet filtering rules.

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 2.2. Kỹ thuật VoIP
4. Gatekeeper:
 A gatekeeper is a management tool that oversees authentication,
authorization, telephone directory, and PBX services.
 Commercial entities implementing VoIP can maintain the billing
information along with the call details in the gatekeeper.
 Although it can be incorporated in the server, generally gatekeepers
are implemented separately to simplify the server operations.

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 2.2. Kỹ thuật VoIP

 How VoIP Works?

-The basic technology of VoIP consists of digitizing
the analog voice and sending it in the form of IP
packets over the Internet or any other IP-based
network.

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2.2. Kỹ thuật VoIP

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 2.2. Kỹ thuật VoIP
 The audio signal as recorded by the input device is sampled at a
very high rate (at least 8000 times per second or more) and
transformed into digital form by an analog-to-digital (A/D)
converter
 The digitized data is further compressed into very small samples
that are collected together into larger chunks and placed into data
packets for transmission over the IP network. This process is
referred to as packetization
 Generally, a single IP packet will contain 10 or more milliseconds of
audio, with 20 or 30 ms being most common
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 2.2. Kỹ thuật VoIP
 There are a number of ways to compress this audio, the algorithm for
which is referred to as a “compressor/ de-compressor”, or simply Codec.
 Many Codecs exist for a variety of applications (e.g., movies and sound
recordings). Most of the Codecs are defined by standards of the ITU-T
 With respect to VoIP, the Codecs are optimized for compressing voice,
which significantly reduces the bandwidth used compared to an
uncompressed audio stream and ensures high quality of VoIP
transmission.
 After binary information is encoded and packetized at the sender end,
packets encapsulating voice data can be transmitted on the network.
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 2.2. Kỹ thuật VoIP

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 2.2. Kỹ thuật VoIP

-As real-time communication is highly sensitive to

loss of information, steps must be taken to minimize
the end-to-end delay and packet loss through
reservation of resources and other techniques.

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 2.2. Kỹ thuật VoIP

-As voice communication occurs in the form of

talkspurts, there is significant “dead” time during
which no speaker is talking. Codecs take advantage
of the silence periods by applying “silence
suppression” techniques that stop transmission
during the idle periods and significantly save the
network bandwidth.
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 2.2. Kỹ thuật VoIP

 A schematic diagram depicting the creation of VoIP

packets after compression, echo cancellation, and silent
suppression is given as:

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 2.3. VoIP Gateway

-Gateways are defined in different ways by different

people. In our context here a gateway is a server; it
may also be called a media gateway
-Media gateways are part of the physical transport layer.
They are controlled by a call control function housed in a
media gateway controller.
-A media gateway with its associated gateway controller is at
the heart of the network transformation to packetized voice
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2.3. VoIP Gateway

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 2.3. VoIP Gateway

 Several of the media gateway functions are listed

here:
 Carries out A/D conversion of the analog voice channel (called
compression in many texts).
 Converts a DS0 or E0 to a binary signal compatible with IP or
ATM.
 Supports several types of access networks including media such as
copper (including various DSL regimes), fiber, radio (wireless), and
CATV cable. It is also able to support various formats found in
PDH and SDH hierarchies. 25
 2.3. VoIP Gateway

 Competitive availability (99.999%).

 Multivendor interoperability.
 It must provide interface between media gateway control device and
the media gateway. This involves one of four protocols: SIP [2],
H.323 [3], MGCP, and Megaco (H.248).
 Can handle switching and media processing based on standard
network PCM, ATM, and traditional IP.
 Transport of voice

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 2.3. VoIP Gateway

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 2.4. Media Gateway Controller

 The gateway controller or media gateway

controller (MGC) carries out the signaling function
on VoIP circuits. Some texts call an MGC a soft
switch even though they are not truly switches but
are servers that control gateways

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 2.4. Media Gateway Controller

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 2.4. Media Gateway Controller
 There are four possible signaling protocol options
between an MGC and gateways.
 ITU-T Rec. H.323. This is employed where all network elements (NEs)
have software intelligence.
 SIP (session initiation protocol) is used when the end devices have
software intelligence and the network itself is without such intelligence.
 MGCP (media gateway control protocol) is another gateway control
protocol.
 Megaco (ITU-T Rec. H.248) is a gateway control protocol applicable
when end devices are without software intelligence and the network has
software intelligence.
 2.5. Chuẩn ITU-T Rec. H.323

-Signaling protocols are used to establish and control multimedia

sessions such as multimedia conferences, telephony, distance learning
-The IP signaling protocols are used to connect software- and
hardware-based clients through a local area network (LAN) or the
Internet
-There are currently two standardized protocols widely deployed in the
market, namely H.323 and SIP. These two protocols provide different
approaches toward attaining the same goal—signaling and control of
multimedia conferences

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 2.5. Chuẩn ITU-T Rec. H.323

a) H.323 is an umbrella specification that covers

many other ITU documents and protocols and is
used for transmitting audio, video, and data across an
IP network, including the Internet.

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 2.5. Chuẩn ITU-T Rec. H.323

-The H.323 standard consists of the components

and protocols:

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 2.5. Chuẩn ITU-T Rec. H.323

b) Elements: H.323 elements play a pivotal role in

maintaining efficient VoIP service

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 2.5. Chuẩn ITU-T Rec. H.323

 Terminal

-H.323 terminals must have a system control unit,

media transmission, audio codec, and packet-based
network interface. Optional requirements include
video codec and user data applications

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 2.5. Chuẩn ITU-T Rec. H.323

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 2.5. Chuẩn ITU-T Rec. H.323

-It provides H.225 and H.245 call control, capability

exchange, messaging, and signaling of commands for
proper operation of the terminal.
-It also formats the transmitted audio, video, data,
control streams, and messages onto network
interface and receives these messages from the
network interface.
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 2.5. Chuẩn ITU-T Rec. H.323
 Gateway
-The H.323 gateway reflects the characteristics of
a switched circuit network (SCN) endpoint and
H.323 endpoint
-It translates into audio, video, and data transmission
formats as well as communication systems and
protocols.
-It also performs compression and packetization of
voice 38
 2.5. Chuẩn ITU-T Rec. H.323


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 2.5. Chuẩn ITU-T Rec. H.323

 Gatekeeper
-An optional entity, the gatekeeper, provides pre-
call and call-level control services to H.323
endpoints.
-Gatekeepers are logically separated from the other
network elements in H.323 environments. It
performs address translation, admissions control,
bandwidth control, and zone management. 40
 2.5. Chuẩn ITU-T Rec. H.323

- The gatekeeper is the most complex component of

the H.323 framework and is responsible for most
network-based services including the direct endpoint
call signaling and gatekeeper-routed call signaling
flows.

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 2.5. Chuẩn ITU-T Rec. H.323

 Multipoint Control Unit (MCU)

MCUs consist of multipoint controller and
multipoint processor. They are involved in issues
related to conferencing

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 2.5. Chuẩn ITU-T Rec. H.323

c) Protocol Suite

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 2.5. Chuẩn ITU-T Rec. H.323

-The H.323 protocol suite supports call admissions,

setup, status, teardown, media streams, and messages
in H.323 systems
-The H.323 protocol suite is split up into three main
areas:

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 2.5. Chuẩn ITU-T Rec. H.323

 Registration, Admissions, and Status (RAS)

Signaling:
+ provides pre-call control in H.323 gatekeeper-based networks.
+The RAS channel is established between endpoints and gatekeepers
across an IP network
+The RAS channel is opened before any other channel is established
and is independent of the call control signaling and media transport
channels

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 2.5. Chuẩn ITU-T Rec. H.323

d)Call Flow
The H.323 call flow comprises of several phases,
such as the initiation phase, capability negotiation
phase, master–slave determination phase, and finally
the dialogue phase.

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 2.5. Chuẩn ITU-T Rec. H.323

 Initializing the call: H.323 initializes the call using SETUP,

ALERTING, and CONNECT messages
 Establishing the control channel : It is done in two
phases: Capability Negotiation Phase, master–slave
determination phase
 Opening media channels: Both Terminals A and B open
media channels for voice and possibly video or data. The
digitized data is carried in several “logical channels.”
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 2.5. Chuẩn ITU-T Rec. H.323

 Dialogue: This is the phase where the caller and

callee actually communicate using voice and/
or video.

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 2.5. Chuẩn ITU-T Rec. H.323

- Gatekeepers are required in scenarios where the

caller or the callee may not be using Internet-based
phones. The gatekeeper is the most complex
component of the H.323 framework and is
responsible for most network-based services.
- There are two call flow methods involving
gatekeepers:
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 2.5. Chuẩn ITU-T Rec. H.323

a) Direct endpoint call signaling: Call signaling

messages are sent directly between the two endpoints

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 2.5. Chuẩn ITU-T Rec. H.323

1. Here, both the endpoints send Admission Request (ARQ)

messages to the gatekeeper.
2. Based on filtering and other admission control features,
the gatekeeper accepts the request by sending Admission
Confirm (ACF) messages or rejects the request sending
Admission Reject (ARJ) messages.
3. If both endpoints are granted admission, then they
exchange setup and connect messages and the call is
established.
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 2.5. Chuẩn ITU-T Rec. H.323

b)Gatekeeper-routed call signaling: Call signaling messages

between the endpoints are routed through the gatekeeper as shown in
Fig. 2.4

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 2.5. Chuẩn ITU-T Rec. H.323

-The gatekeeper has centralized control over the entire

duration of the call. Unlike the previous scenario, in this
case, even the setup and connect messages are routed
through the gatekeeper. However, there may be a
modification to this model where the RTP flows may be
established directly between the two endpoints in order to
reduce the latency

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 2.6. Giao thức SIP
 SIP is an application-layer control protocol that
can establish, modify, and terminate multimedia
sessions (conferences) such as Internet telephony
calls with simple call flows and messages.

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 2.6. Giao thức SIP

-SIP supports the five facets of establishing and

terminating multimedia communications:

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 2.6. Giao thức SIP
a) SIP Actors: refer to all those entities who are
involved in the creation, management, and
termination of SIP sessions.
 User Agents
A. User agent client (calling party): is a logical
entity that creates a new request and then uses
the client transaction state machinery to send it.
The role of UAC lasts only for the duration of that
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transaction.
 2.6. Giao thức SIP

B. User agent server (called party): is a logical entity that

generates a response to a SIP request. The response accepts,
rejects, or redirects the request for the entire duration of the
transaction.
 Servers
A. Proxy Server: It is an intermediary entity that acts as
both a server and a client for the purpose of making
requests on behalf of other clients. A proxy server
primarily plays the role of routing
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 2.6. Giao thức SIP

B. Redirect Server: is a server that generates 3xx

responses to requests it receives, directing the client
to contact an alternate URI.
C. Registrar: is a server that accepts register requests
and places the information it
receives in those requests into the location service
for the domain it handles.
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 2.6. Giao thức SIP

b) SIP Structure: SIP is structured as a layered

protocol

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 2.6. Giao thức SIP

c) SIP Message Type

-SIP is a text-based protocol and uses the ISO 10646
character set in UTF-8 encoding (RFC 3629)
- A SIP message is either a request from a client to a
server, or a response from a server to a client.
- The syntax is as follows:
generic-message = [start-line] + [message-header]
+ [CRLF] + [message-body]. 60
 2.6. Giao thức SIP

A. Requests
-SIP requests are distinguished by having a
Request-Line for a start-line.
-A Request-Line contains a method name, a
Request-URI, and the protocol version separated by
a single space (SP) character. The Request-Line ends
with CRLF
-The syntax is Method Request-URI SIP-Version. 61
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 2.6. Giao thức SIP

B. Responses
-The syntax is SIP-version Status-Code Reason-
Phrase.
-SIP responses are of two types:
• Provisional (1xx class)—provisional responses are used by
the server to indicate progress, but they do not terminate
SIP transactions.
• Final (2xx, 3xx, 4xx, 5xx, 6xx classes)—final responses
terminate SIP transactions. 63
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 2.6. Giao thức SIP

d) SIP Call Flows

A. SIP Call Flow with Proxy Server

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 2.6. Giao thức SIP
1. The proxy server accepts the INVITE request from the client.
2. The proxy server identifies the location by using the supplied
addresses and location services.
3. An INVITE request is issued to the address of the location
returned.
4. The called party user agent alerts the user and returns a success
indication to the requesting proxy server.
5. An OK (200) response is sent from the proxy server to the calling
party.
6. The calling party confirms receipt by issuing an ACK request,
which is forwarded by the proxy or sent directly to the called party.
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 2.6. Giao thức SIP

B. SIP Call Flow with Redirect Server

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 2.6. Giao thức SIP
1. The redirect server accepts the INVITE request from the calling
party and contacts location services with the supplied information.
2. After the user is located, the redirect server returns the address
directly to the called party. Unlike the proxy server, the redirect server
does not issue an INVITE.
3. The user agent sends an ACK to the redirect server acknowledging
the completed transaction.
4. The user agent sends an INVITE request directly to the address
returned by the redirect server.
5. The called party provides a success indication (200 OK), and the
calling 69
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