University

ISAM Voice Operator

Managed with 5520 AMS

Section B: VoIP Introduction
Alcatel-Lucent University Antwerp
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Alcatel-Lucent University Antwerp

During class please switch off your mobile, pager or other that may interrupt.

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Course objectives
At the end of this section, you will be able to
Understand the VoIP background of ISAM Voice

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Table of contents
Voice over IP (VoIP) .
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MEGACO / H.248
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Session Initiation Protocol (SIP) .

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University
Voice over IP (VoIP)

Alcatel-Lucent University Antwerp

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Network Models for telephony over IP

MGC Model, defined by ITU-T:

SIP model defined by IETF:

Media Gateway (MG)

Media Gateway Controller (MGC)
Signalling Gateway (SG)

H.248/MEGACO
Used with Telephony NGN
support legacy terminals
Master-Slave
Intelligent server (MGC)
Intelligence Centralized in MGC

Voice Gateway (VGW)

Call Session Controller (CSC)
Terminal

SIP
Used with Multimedia NGN
IP Phone, Multimedia terminals
Peer-to-peer
Intelligent endpoint
Intelligence decentralized

Two international organisations as the IETF and ITU propose two different network models, each
one having its own set of features and protocols.
IETF: Internet Engineering Task Force
ITU: International Telecommunication Union

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Data Plane: Coding techniques

ISAM-V supports G711, G723 and G729

Standard C ompression Rate (kbps)

G711
P CM
64
G729
CS -ACE LP
8
G723
MP -MLQ
5,3
G726
ADP CM
16/24/32
G728
LD-ACE LP
16
PCM:
ADPCM:

MOS
4,2
4
3,7
4
4

Delay (ms)
0,75
10
30
1
3-5

64 Kbps, without compression; it is the quality reference

three levels of compression (32 Kbps, 24 Kbps, 16 Kbps)
No quality degradation observed

LDCELP and CS-ACELP: designed for human speech

> PCM - Pulse Code Modulation is the standard for voice quality expected within a PSTN. PCM
operates at 64 kbps, no compression methods being available and thus there is no possibility of
saving bandwidth.
> ADPCM - Adaptive Differential Pulse Code Modulation
> LDCELP - Low-Delay Code-Excited Linear-Prediction (LDCELP)
> CS-ACELP Conjugate-Structure Algebraic-Code-Excited Linear- Prediction
> MP-MLQ - Multipulse Maximum Likelihood Quantization
> MOS - In voice communications, particularly Internet telephony, the Mean Opinion Score (MOS)
provides a numerical measure of the quality of human speech at the destination end of the circuit.
To determine MOS, a number of listeners rate the quality of test sentences read aloud over the
communications circuit by male and female speakers. A listener gives each sentence a rating as
follows: (1) bad; (2) poor; (3) fair; (4) good; (5) excellent. The MOS is the arithmetic mean of all the
individual scores, and can range from 1 (worst) to 5 (best).
> ISAM-V support:

Codec

Packetisation times

Bit rate

G.711

5, 10 and 20 ms

64 kbps

G.723.1 (*)

30 ms

6.3 and 5.3 kbps

G.729

10 and 20 ms

8 kbps

(*) G723 supported from R1.1 onwards

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RTP/RTCP
RTP = Real-time Transport Protocol
for delivering audio and video over IP networks

RTCP = Real-time Transport Control Protocol

out-of-band control of an RTP flow
RTP
RTP
Header

RTP/RTCP
UDP
IP
802.1Q
VLAN = Voice
Ethernet MAC

Content:
audio, video, etc.
RTP Packet : signal
SOURCE

DESTINATION
RTCP Packet : feed-back
RTCP
RTCP
Header
Receiver reports

> The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering
audio and video over IP networks. It was developed by the Audio-Video Transport Working
Group of the IETF and first published in 1996 as RFC 3550.
RTP does not have a standard TCP or UDP port that it communicates on. The only standard that it
obeys is that UDP communications are done via an even port and the next higher odd port is used
for RTP Control Protocol (RTCP) communications. RTP only carries voice/video data.
> RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). It is
defined in RFC 3550
RTCP stands for Real-time Transport Control Protocol, provides out-of-band control information for
an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not
transport any data itself. It is used periodically to transmit control packets to participants in a
streaming multimedia session. The primary function of RTCP is to provide feedback on the
quality of service being provided by RTP.
It gathers statistics on a media connection and information such as bytes sent, packets sent, lost
packets, jitter, feedback and round trip delay. An application may use this information to increase
the quality of service perhaps by limiting flow, or maybe using a low compression codec instead of
a high compression codec. RTCP is used for QoS reporting.

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RTP/RTCP The overall picture

Using codec G711 with 20 ms packetisation, the overall
bandwidth (Ethernet layer) = 97 kbps / call
TDM voice flow at 64 kbps
TDM flow is packetized
CODEC (voice compression), silence suppression...
RTP header added (sequence, timestamp)
UDP header added (src and dest port)
IP header added (src and dest address)
Ethernet header and FCS Trailer

> For IP: Suppose we have 20 ms packetization time using G.711, so every IP packet:
20ms/0.125ms = 160 bytes, 12 bytes RTP header, 8 bytes UDP header, 20 bytes IP header, total
= 200 bytes/packet
> For Ethernet, we have Preamble 7 bytes, frame delimiter 1 byte, destination MAC 6 bytes, source
MAC 6 bytes, VLAN 4 bytes, type field 2 bytes, FCS 4 bytes, inter-frame gap 12 bytes, total = 42
bytes of overhead or a total of 242 bytes / packet
> Number of packets per second = 1000 / 20 = 50 packets per second
> Ethernet Bandwidth = 50 * 242 * 8 = 96,8 kbits/sec
(TDM: 64 kbps => 8000 samples/sec, each sample = 1 byte => 8000*8 = 64kbps)

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University
MEGACO/H.248

Alcatel-Lucent University Antwerp

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MEGACO/H.248 introduction
Signalling protocol
between Media Gateway (MG) and Media Gateway Controller (MGC)

Media Gateways are controlled by the MGC

All Point-to-point connections from MGC to Media Gateways (MG)
SS7
network

SIGTRAN

SG

MGC

TG

IP

Fax

MEGACO

IP

MEGACO

PSTN

SIGTRAN

IP
backbone
IP

RTP

IP

AG

Analog/ISDN
phone

Analog/ISDN
phone

IP

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> 2 types of Media Gateways: TG = Trunk Gateway / AG = Access Gateway

> MEGACO (a contraction of "Media Gateway Controller") is a signalling protocol, used between a
Media Gateway and a Media Gateway Controller (also known as a Call Agent or a Soft Switch) in
a VoIP network. It defines the necessary signalling mechanism to allow a Media Gateway
Controller (Call agent) to control gateways in order to support voice/fax calls between PSTN-IP or
IP-IP networks.
> MEGACO is a protocol sponsored by IETF and ITU, used within the Media Gateway Controller
(MGC) to control Media Gateways (MGs)
> This protocol permits the MGC to indicate to the Gws to send and receive information streams
to/from specific addresses, generating tones and modifying the configuration. It also permits the
GWs to send reports to the MGC regarding events detected such as off-hook/on-hook states,
digit detection etc.
> All in all, MEGACO allows the call control function to be implemented using devices external to
the gateways proper, such as the Media Gateway Controller.
> MEGACO or MGCP: (MGCP = Media Gateway Control Protocol)
MGCP
= IPDC ( used by Level 3 , Alcatel , Lucent , .. ) + SGCP ( used by Cisco and Bellcore)
Result = MGCP ( IETF RFC2705 ; oct 1999 )
MEGACO
IETF RFC 2885 , August 2000
same as H.248 ( few details different )
> MEGACO is a protocol operating on the UDP transport layer and its ITU equivalent is the H.248
protocol.

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MEGACO/H.248 introduction
Designed for Multi Party Calls ( Point To Multi Point )
Runs over any kind of access ( IP , ATM , SDH , )
Difference MEGACO (IETF) and H.248 (ITU-T)
MEGACO : text
H.248 : ASN.1

ALL Call Control in Media Gateway Controller

NO Call control in Media Gateway
SIP, e.g. has distributed Call Control (In IP phone, Proxy
server...)

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> Unlike MGCP , that was only designed for two party calls , MEGACO has been designed also for
Multi Party Calls ( example Conference calls ). That is one of the reasons that MEGACO is used
more often than MGCP
> MEGACO can run over any kind of accesss ( IP , ATM , SDH , )
> There are two versions : MEGACO , which has been developed by IETF and H.248 , which has
been designed by ITU_T . There is however little difference between both protocols . The only
difference is that MEGACO is text based and H.248 is ( like all ITU-T protocols ) is ASN.1 based
> It is VERY IMPORTANT to know that MEGACO means :
NO Call Control in Media Gateway
ALL Call Control in Media Gateway Controller
This is the big difference between MEGACO and SIP:
in SIP , the call control is distributed over the diferent elements ( Ip Phone , Proxy
Server , .. )
> Abstract Syntax Notation One (ASN.1) is a formal language for abstractly describing messages
to be exchanged among an extensive range of applications involving the Internet, intelligent
network, cellular phones, ground-to-air communications, electronic commerce, secure electronic
services, interactive television, intelligent transportation systems, Voice Over IP and others.

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MEGACO/H.248 introduction (1)

The Media Gateway Controller (MGC)

Performs the connection management function
Allocation of bearer terminations and media resources in MG
Specifies events to be monitored
Defines Media transport like compression
Collects statistics from the MG
Controls supplementary services and call routing
Secure billing functions

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MEDIA GATEWAY CONTROLLER

Performs the connection management function
Controls the allocation and de-allocation of bearer terminations and media resources in the
Media Gateway for each call established
Specifies at each phase of the call the events to be monitored or the signals to be applied by
the Media Gateway
Defines the media transformation parameters (echo cancellation, compression, silence
suppression, etc) for each media stream that is part of a call.
Collects statistics cumulated by the Media Gateway, such as volume of content carried, call
duration, QoS statistics, etc.
Controls supplementary services and call routing.
Secure billing functions.

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MEGACO/H.248 introduction (2)

The Media Gateway (MG)
Converts media provided in one type of network to the format
required in another type of network

The Distributed Access Media Gateway

Provides direct interfaces to POTS/ISDN lines
Performs compression, echo cancellation, silence suppression,
packet loss concealment, etc...
Additional support for basic and supplementary services
-

call progress tones

detection of tones and DTM digits
processing of digit maps
detection of line events
insertion of data signals for CLI services
control of line feed and metering signals,...

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University
Session Initiation Protocol (SIP)

Alcatel-Lucent University Antwerp

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SIP introduction (1)

Signalling protocol
Between end-user terminals such as Voice Gateways or IP phones
and the Call Session Controller

Voice Gateways are intelligent and independent devices

They are seen as end-user terminals (with many lines) and must
register each line (user) with the CSC before making calls
IP phone

SS7
network

SG
SIGTRAN

PSTN

SIP

CSC

VGW

Fax

IP

IP
backbone
IP

RTP

IP

SIP

IP

SIP

IP
IP

VGW

Analog/ISDN
phone

Analog/ISDN
phone

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SIP Introduction (2)

A Call Session Controller (CSC) is a combination of
a SIP Registrar (for keeping terminal registrations),
a SIP Proxy Server (for routing calls) and
a service platform (for triggering service execution on
Application Servers)

The SIP model uses distributed call control

End-user terminals have (limited) call control (e.g. call
forwarding)
However, intelligent routing of calls and service activation is
centralized in the CSC

In IMS the CSC is split-up in separate components focussing on

a certain task (see next slides)

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IMS key components (1)

The Call Session Control Function (CSCF) establishes,
monitors, supports and releases multimedia sessions and
manages the user's service interactions. The CSCF can act as
Proxy CSCF (P-CSCF), Serving CSCF (S-CSCF) or Interrogating
CSCF (I-CSCF):
The P-CSCF is the first contact point for the ISAM Voice within
the IM subsystem (IMS)
The S-CSCF actually handles the session states in the network
The I-CSCF is mainly the contact point within an operator's
network for all IMS connections destined to a subscriber of that
network operator, or a roaming subscriber currently located
within that network operator's service area.

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IMS key components (2)

The Home Subscriber Server (HSS) is a master user database
that supports the IMS network entities that actually handle
calls. It contains the subscription-related information (user
profiles), performs authentication and authorization of the
user, and can provide information about the user's physical
location.
Interconnection with legacy PSTN networks is guaranteed at
the signaling level via the Signaling Gateway Function (SGF)
(transport) and the Media Gateway Control Function (MGCF)
(call/service control). Interconnection at the media level is
provided by the Trunk Media Gateway Function interface (TMGF).

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IMS domains and user identities

Each IMS operator has his own domain name
This domain name is used in registering and in making calls

Each user has two identities:

A private identity, used to authenticate, like an account name
Only known between user and operator

A public identity, like a telephone number or an e-mail address

Can be found in public directories

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www.alcatel-lucent.com

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