ISAM Voice Product Overview

NokiaEDU
ISAM Voice Product Overview

TAC05021W
Version 5.7 - Edition 1
© NOKIA 2017. ALL RIGHTS RESERVED.

COURSE CONTROL Glossary
and
Side Resources
Panel
Next and
Control
Previous
bar
Buttons
Course Objectives
• Describe the VoIP background of ISAM Voice
• Explain how VoIP is transmitted
• Distinguish between MEGACO and SIP protocol
• Explain how the ISAM Voice Package fits into the

ISAM equipment practice
• Describe the concepts of the ISAM SIP and MEGACO

Voice Architecture
• Describe narrowband line testing

Course Outline
1. Course Introduction
2. ISAM Voice over IP overview
3. ISAM Voice Equipment practice
4. ISAM Voice Architecture – MEGACO
5. ISAM Voice Architecture – SIP
6. ISAM Voice NBLT

NokiaEDU
ISAM Voice over IP overview
Version 5.3 - Edition 2

OBJECTIVES

VOICE OVER IP CONCEPTS

PACKET SWITCHING VS CIRCUIT SWITCHING
Circuit switching Packet switching
 Constant 64 Kbps rate. It does not matter  Burst transmission. Bits are only transmitted
if information is being transmitted when there is information to be sent
(voice) or not (silence)  Each packet needs a routing label indicating
 Once the voice circuit is established, its destination. Each packet has to be routed
voice goes always through it and knows and switched
its destination  There is no physical circuit “dedicated” to
 Each Circuit handles only one each conversation
conversation  Resources are shared among all conversations
 Resource usage is not optimized  Delays are variable in time (jitter)
 Delay is fixed and stable (no jitter)
Circuit Switching Network Packet Switching Network
64 Kbps
circuit
VoIP facilitates Network Convergence by adding voice services support to IP Networks

VoIP TRAFFIC TYPES: VOICE AND SIGNALING
Voice traffic Signaling traffic
 Voice data that has been sampled and  This traffic contains control and signaling
coded using a specific CODEC information among the different network
elements involved in the VoIP service
 Voice traffic uses RTP as underlying delivery.
protocol
 Signaling traffic uses H.248 or SIP as main
 Voice data is typically exchanged protocols.
directly between the end point  Signaling data is exchanged between the core
devices in the network (unless forced and access elements in the VoIP network in
by the service provider) order to control the VoIP services.
Core (NGN/IMS)
Core (NGN/IMS)
SIP/H.248 SIP/H.248
IP Network
RTP (Voice traffic)
IP Network
Access Gateway/ONT Access Gateway/ONT Access Gateway/ONT Access Gateway/ONT

CODEC RELATED CONCEPTS
Codec characteristics
Voice Codecs are compression algorithms Bit Rate

designed to reduce the transmission
bandwidth of digital voice streams. Packetization Interval
(ptime)
Algorithm Delay
"Waveform“:
 Based on sampling and coding the wave
Look-Ahead Delay
 G711 and G726
“Vocoders”:
 Based on coding parameters to allow voice synthesis Codec Quality (MOS)
 G729
Coding Algorithm
VOICE TRAFFIC: CODING TECHNIQUES
Standard Compression Rate (kbps) MOS Delay (ms)
G711 PCM 64 4.4 0.75
G726 ADPCM 16/24/32 4 1
G728 LD-CELP 16 4 3-5
G729 CS-ACELP 8 4 10
G723.1 MPC-MLQ 6.3 3.9 37.5
G723.1 ACELP 5.3 3.6 37.5
PCM: 64 Kbps, without compression; it is the quality reference
ADPCM: three levels of compression (32 Kbps, 24 Kbps, 16 Kbps)
 No quality degradation observed
2 compression techniques for G.723.1:
6.3 kbit/s (using 24 byte frames) using a MPC-MLQ algorithm (MOS 3.9)
5.3 kbit/s (using 20 byte frames) using an ACELP algorithm (MOS 3.62)
SILENCE REMOVAL: VAD AND CNG
One important capability of VoIP technology is the ability to reduce
the transmission bandwidth usage by detecting and removing the
periods of silence present in every voice conversation getting a
bandwidth saving of up to 60%.
Voice Activity Detection (VAD):

Comfort Noise Generation (CNG):
Identifies the silence during speech and
stops data transmission during them. Is the background noise during periods
of silence in a call.
This allows the user to check the call is

not dropped.
VAD module sends Silence Insertion

Descriptor (SID) to generate 'realistic'
acoustic background noise.

ISSUES AFFECTING VOICE QUALITY IN VoIP
How to minimize impact of main factors which affect VoIP Quality
Codec Intrinsic
Codec's Fidelity Quality
Efficiency (% Packet Loss

packets correctly Concealment
received)
Network Delay
Delay Improvements
Jitter Jitter Buffer
Echo Echo Cancel

VOICE TRAFFIC: RTP AND 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
Header
Content:
Audio,video, …
RTP Packet: Signal
SOURCE DESTINATION
RTCP Packet: Feed-back
RTCP Header
Receiver
reports
RTP/RTCP: THE OVERALL PICTURE
Using codec G711 with 20 ms packetization, 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

VOICE ARCHITECTURE AND PROTOCOLS

VoIP NETWORK MODELS: IMS PES VS IMS PSS
ITU IMS supports both models
PES: PSTN Emulation PSS: PSTN Simulation

 Emulation service provides the same  Provide “PSTN-like” services to users
service offerings and user experience  Based on the usage of new IP
as traditional TDM switches “Terminals”
 No impact on User equipment and  Includes terminal adapters
interfaces  Supported on IMS network architecture
 Two approaches for PES:
• NGN Call Server(Softswitch)
• IMS-based

VoIP PROTOCOLS
H.248 SIP
Used with telephony Used with multimedia
Support legacy terminals IP Phone, multimedia terminals
Master – Slave Peer-to-Peer
Intelligent server (MGC) Intelligent endpoint
Intelligent centralized in MGC Intelligent decentralized
POTS/ISDN Only POTS

VoIP ARCHITECTURE: MAIN FUNCTIONAL ELEMENTS
IP Network
VoIP PSTN Network
IP Network
Access POTS phones VoIP Voice Trunk
Gateway Trunk
Media
Access/Voice Gateway: Gateway PSTN switch
Performs the conversion
between voice channels of its (Trunk) Media Gateway:
directly connected terminals Is the interface between PSTN
(ex: POTS) and VoIP media and VoIP networks. TMG can
Signaling
streams. Gateway typically manage a large
number of digital circuits.
PSTN Network
IP Network
VoIP SS7
SIGTRAN PSTN switch
Signaling Gateway:
Makes conversion between
PSTN and VoIP Signaling.

VoIP ARCHITECTURE: MAIN FUNCTIONAL ELEMENTS
[CONT.]
Announcements
Tones, IVR IP Network
Conference Bridges
IP Network
Media
Resource
Server Telephony Application Server (TAS):
It provides the telephony service logic
Media Resource Server (MRS): and control on IMS networks.
Provides additional resources like
announcements, tones.
IP Network
Media Gateway Controller (Softswitch):

It is the “brain” of the NGN network
implementing the call logic and control.

ACCESS AND VOICE GATEWAYS
Access and voice gateways:
 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
• etc. Analog/ISDN
IP IP
phone
backbone AG
Analog/ISDN
phone
NGN NETWORK ARCHITECTURE
Media Gateways are controlled by the MGC:
 All Point-to-point connections from MGC to Media Gateways (MG)
MGC SS7
(Softswitch) NGN
H.248/
SIGTRAN
SG PSTN
Media
Resource H.248
E1 network
Server H.248
TGW
H.248/ ISDN IUA
(SIGTRAN) PSTN switch
H.248
Access
Gateway H.248 Signaling protocol between Media
ONT Gateway (MG) and Media Gateway
ISDN
ISDN
NT
NT Controller (MGC)
POTS
ISDN POTS
Next Generation Networks represents the natural evolution from TDM into the VoIP
world, characterized by a centralized service control from the Softswitch.
SIGNALING PROTOCOLS: H.248 (AKA MEGACO)
Designed for multi party calls (Point to multi point)
Runs over any kind of access (such as Ethernet, ATM, SDH)

MEGACO SG 16
SGC
IPDC
July/98 P Aug/98
MGC MDC
Oct/98 P P Nov/98
MEGAC MGC
Original MEGACO (IETF) evolved O
Apr/99 P
to H.248 (ITU-T)
H.248
June/2003 March/2002
ALL call control in the media gateway controller

NO call control in the media gateway
SIGNALING PROTOCOLS: H.248 (AKA MEGACO) [CONT.]
H.248 is a signaling protocol, used between an access device (Media
gateway) and a Media Gateway Controller (MGC) in a VoIP network.
H.248 Characteristics: MGC
(Softswitch)
 H.248 works in Master-Slave mode. MGC is Master
and MG is the Slave H.248 H.248
 H.248 is typically used on NGN where voice
service logic is centralized in the Softswitch IP Network
(MGC). RTP (Voice traffic)
 Both UDP and TCP are allowed as underlying

transport protocol, being UDP the most used.
MG
 ITU-T H.248 supports both Binary and Text ASCII MG
encoding. Session Description Protocol (SDP)

 H.248 standard is structured in packages. Each SDP is used for negotiation between
package defines a specific set of properties, end points of media type, format,
events and signals. A network element H.248 and all associated properties
SDP information is embedded inside
compliancy is described by the set of supported H.248 messages.
packages.
IMS ACCESS NETWORK ARCHITECTURE
Voice IMS Application Server
Gateway
(UA) SBC
SIP P-CSCF
SIP
I/S-CSCF
SIP AGCF MRF
ONT BGCF
(UA)
SIP
SIP, signaling protocol for IMS
POTS
SS7
H.248 supported by AGCF
H.248 H.248 E1
MGCF PSTN
H.248 /IUA
TGW/MGW
network
(SIGTRAN)
Access
Gateway
ONT ISDN
ISDN
NT
POTS NT
ISDN
IP Multimedia System (IMS) Networks represents an advanced architecture for the introduction of new
services and the convergence of fixed and mobile worlds.
SIGNALING PROTOCOLS: SIP
Session Initiation Protocol (SIP) is a signaling protocol, typically
used on IMS architectures.
SIP Characteristics: Application
 SIP is an IETF-defined signaling protocol It Server
is a text-based protocol. I/S-CSCF
IMS
 SIP is typically used on IMS architectures,
either distributed at the end user devices P-CSCF
(UAs) or centralized at the Application
Server (AS) level.
 SIP relies on other protocols, mainly SDP.
 SIP runs on any transport protocol (UDP, SIP SIP
TCP, TLS, SCTP), being UDP the most used. IP Network
Session Description Protocol (SDP) RTP (Voice traffic)
SDP is used for negotiation between end points of

media type, format, and all associated properties UA
SDP information is embedded inside SIP messages. UA

IMS ELEMENTS
IMS is a layered model:
AS
Application layer: The application layer
contains servers and top level applications MGCF
needed to deliver services.
HSS S-CSCF BGCF
Control Layer: Here is the core network
on IMS to deliver main functionality.
P-CSCF Media PSTN

User layer: This is the portion of the IMS Services Interface
architecture through which the overall PCF
network is accessed.
User equipment: The user equipment is

the part of the IMS architecture that Subscriber To PSTN
resides with the user - it is the endpoint.
Simplified view of the functional IMS architecture

IMS ELEMENTS [CONT.]
The Call Session Control Function
(CSCF) establishes, monitors, supports
and releases multimedia sessions and AS
manages the user's service interactions:
MGCF
 The Proxy CSCF, P-CSCF is the first
contact point for the Voice subscribers. HSS S-CSCF BGCF
 The Serving CSCF, S-CSCF actually
handles the session states in the
network.
 The Interrogating CSCF, I-CSCF is P-CSCF Media PSTN
mainly the contact point for all IMS Services Interface
connections destined to a subscriber of PCF
that network operator.
The Home Subscriber Server (HSS) is a
user database. It contains: Subscriber To PSTN
 Subscription-related information (user
profiles) Simplified view of the functional IMS architecture
 User’s authentication and authorization
 User’s physical location © NOKIA 2017. ALL RIGHTS RESERVED.
IMS ELEMENTS [CONT.]
Interconnection functions: AS
 To PSTN via Signaling Gateway MCGF
Function (SGF) (transport) and
the Media Gateway Control HSS S-CSCF BCGF
Function (MGCF) (call/service
control).
 Media Services at the media Media PSTN
P-CSCF
level is provided by the Trunk Services Interface
Media Gateway Function PCF
interface (T-MGF).
Subscriber To PSTN
Simplified view of the functional IMS architecture

VOICE ACCESS EVOLUTION
TDM/V5 NGN IMS
Media
Gateway
PSTN IP H.248 Controller IP SIP
Network Network Network IMS core
Service H.248 AGCF

Local Service SIP
Exchange TGW Router
Router
Service
Router SIP
E1/STM-1 V5.2 FE/GE H.248
FE/GE
ISDN
NT ONT ONT
ISDN
ISDN NT
ISDN NT POTS POTS POTS
POTS
ISDN
POTS
Smooth PSTN to NGN/IMS migration
VOICE INTEROPERABILITY

INTEROPERABILITY CONCEPTS
Voice Interoperability is the ability of the voice access system and the
core network to work together in order to fulfil the customer
requirements: Basic call, Supplementary Services, Data and Fax calls.
Voice Interoperability is a complex task as it requires

to adapt the access and core equipments to the
customer legacy voice requirements.
Interoperability is NOT the same as protocol

compliance, as standards are open to multiple
interpretations and implementations for
supplementary services.

INTEROPERABILITY BLOCKS
H.248/SIP <->
Softswitch Sx Subscriber Line
IMS-AGCF/MGC Interworking
H.248/SIP Subscriber line
Signaling Specification Interface
Specification Specification
H.248/SIP  Subscriber
Line Interworking
H.248/SIP signaling Subscriber Line

signaling
POTS
IP AGW / VGW
AGW /
SCTP/IUA
network VGW FAX
ISDN BRI
ISDN
Access Node ISDN PRI

SUPPLEMENTARY SERVICES (SS) OVERVIEW
Voice Supplementary Services are all those end user services that provide added value
beyond the basic call handling, as for example Call Forwarding, Call Waiting, CLIP, etc.
The support of Supplementary Services Common Supplementary Services

depends in a great measure on the
Calling Party Identification Services
scenario “dictated” by the Call Server,
ISAM is the “slave”:
Conference Services
- Fully dependant if Megaco
- Highly dependant if SIP “tightly
Direct Connect services
coupled”
- Less dependent if SIP “loosely
Call Forwarding Services
coupled”
Call Hold and Call Waiting
Implementation of supplementary Services
services is highly dependant on the
service provider and the core network Call Barring Services
with a clear historical legacy coming
from the PSTN network. Call Transfer Services

VOICE FEATURES

BASIC CALL HANDLING FUNCTIONALITIES
Main functionality supported by both the MEGACO and SIP ISAM Voice and SIP for
POTS interfaces:
• Interfacing at subscriber's side according to the Z POTS interface
• Signalling events processing
• Voice activity detection, silence suppression, comfort noise, and packet loss concealment
• DTMF/pulse detection/processing
• Ring generation
• Tone generation; towards subscriber side and network side (Ring-back tone)
• Howler tone generation
• Configurable jitter buffer: Adaptive or fixed size (per call)
• Metering pulse generation
• Near end echo cancellation (G.168, G165 and none)
• Fax/modem detection and transparent fax and modem service (V.150)
• Support of DTMF and fax/modem tones Relay (RFC 2833)
• T.38 Fax Relay
BASIC CALL HANDLING FUNCTIONALITIES [CONT.]
Besides generic Voice functionalities mentioned before, ISAM Voice is full IP and
therefore supports following CODECs:
• G.711 (with 10 ms, 20 ms or 30 ms packetisation time) and either
• G.729 A/B (with 10 ms, 20 ms or 30 ms packetisation time)
• G.723.1
• G.168 (echo cancellation), and,
• T.38 (Group 3 facsimile for VoIP)

ISDN BASIC RATE VOICE SERVICE
Supported by ISAM Voice MEGACO only
ISAM Voice provides (besides basic functionalities mentioned before):
• Termination of the ISDN BRI physical U interface (ITU G.961)
• Q.921 protocol termination
• Q.931 protocol relay via SIGTRAN
• Termination of Q.931 SAPI & TEI. ISAM Voice responsible for TEI
assignment in conjunction with Call Server (ASP).

SIP – TIGHTLY AND LOOSELY COUPLE MODES
Tightly/Loosely coupled concept applies to Tightly/Loosely Coupled Modes
those services that require the subscriber
interaction by pressing the hook-flash key
during a call (3WC, CH, CW, CT) and IMS core IMS core
refers specifically to how the hook-flash key Service logic
is handled on the service logic
implementation:
hook-flash hook-flash
• On Tightly Coupled, hook-flash is Report local handling
reported towards the core. The
service logic is implemented at the Call
Call state
core in a centralized mode. state
• On Loosely Coupled, hook-flash is
internally processed. The service logic Service logic
is implemented at the access node.
SIP Tightly or Loosely Coupled defines how
those Supplementary Services are
handled, by Access or by Core. ISAM Voice Loosely coupled
Tightly coupled
supports both models. © NOKIA 2017. ALL RIGHTS RESERVED.
FAX AND MODEM SERVICES
Transmission of Fax and Modem data over VoIP networks require a
proper adaptation of the end-to-end network to avoid data loss.
NGN/IMS
Fax and Modem Transmission:
• Transmission of fax and modem over VoIP PSTN H.248/SIP
IP
networks require special handling to avoid the Network Network IP
Local
data distortion that may introduce the Exchange Network
Service
mechanisms designed to transmit voice at low TGW Router Service
Router
bit rates (for example voice codecs, echo
cancellation, jitter buffers, VAD, and so on). H.248/SIP
• The mechanisms implemented at the access FE/GE
network are based on the detection of the VBD/T38
fax/modem initially transmitted tones (ANS, VBD/T38
CNG).
ANS, /ANS
• Two main mechanisms exist for the data Modem
CNG, CED
transmission: Voice Band Data (VBD) (For Fax CNG, CED

Modem
Fax and Modems) and T.38 protocol (Fax
only).
© NOKIA 2017. ALL RIGHTS RESERVED. Fax Fax
FAX AND MODEM SERVICES [CONT.]
Transparent fax and modem calls:
• Processed as if they were regular voice calls (based on the usage of G.711 codec) but:
- No echo cancellation, compression and silence suppression, and a fixed-size jitter buffer (i.e. disable the
"adaptive" jitter buffer)
• High and constant bandwidth
T.38, real time fax over IP:

• T38 Fax Relay consist on decoding the modulated fax native T.30 data received from the
fax machine and recode it into T.38 data for transmission over the VoIP network (so no
digitalized audio is transmitted but the original fax data)
• Fax call identified by means of optional CNG, and/or ANS tones
• The use of T.38 mode can be instructed by Call Server or “agreed” end-to-end
T.38 Fax Relay
Real-Time protocol
Increased robustness against packet loss/jitter
Fax Error Correction Capabilities
NIAT-A: SIP ISDN PRI
• NIAT-A card is introduced to support both SIP and ISDN PRI (integrated SIP
ISDN PRI) which mainly deals the basic functionality of DRCI.
• PRI = Primary Rate Interface
• DRCI = Data Rate Control Interface
• NIAT-A has 8 E1 ISDN PRI interfaces/ports for ETSI or 8 T1 ISDN PRI

interfaces/ports for ANSI.
• The SIP ISDN PRI provides support for the shelves 7302FD, 7330FD, 7356 and
7360 with NT boards NANT-A/D/E, AGNT, NRNT-A and FANT-F.

Module Summary
Now that you have completed this course, you should be able to:
Describe the VoIP background of ISAM Voice
Explain how VoIP is transmitted
Distinguish between MEGACO and SIP protocol
Understand the main key components in an IMS platform

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ISAM Voice Product Overview

© NOKIA 2017. ALL RIGHTS RESERVED.

• Describe the VoIP background of ISAM Voice

• Explain how VoIP is transmitted

• Distinguish between MEGACO and SIP protocol

• Explain how the ISAM Voice Package fits into the

• Describe the concepts of the ISAM SIP and MEGACO

• Describe narrowband line testing

© NOKIA 2017. ALL RIGHTS RESERVED.

© NOKIA 2017. ALL RIGHTS RESERVED.

© NOKIA 2017. ALL RIGHTS RESERVED.

© NOKIA 2017. ALL RIGHTS RESERVED.

© NOKIA 2017. ALL RIGHTS RESERVED.

Circuit Switching Network Packet Switching Network

VoIP facilitates Network Convergence by adding voice services support to IP Networks

Access Gateway/ONT Access Gateway/ONT Access Gateway/ONT Access Gateway/ONT

© NOKIA 2017. ALL RIGHTS RESERVED.

Voice Codecs are compression algorithms Bit Rate

Voice Activity Detection (VAD):

This allows the user to check the call is

VAD module sends Silence Insertion

© NOKIA 2017. ALL RIGHTS RESERVED.

Efficiency (% Packet Loss

Jitter Jitter Buffer

Echo Echo Cancel

RTCP = Real-time Transport Control Protocol:

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

© NOKIA 2017. ALL RIGHTS RESERVED.

PES: PSTN Emulation PSS: PSTN Simulation

© NOKIA 2017. ALL RIGHTS RESERVED.

Used with telephony Used with multimedia

Support legacy terminals IP Phone, multimedia terminals

Master – Slave Peer-to-Peer

Intelligent server (MGC) Intelligent endpoint

Intelligent centralized in MGC Intelligent decentralized

POTS/ISDN Only POTS

© NOKIA 2017. ALL RIGHTS RESERVED.

© NOKIA 2017. ALL RIGHTS RESERVED.

Media Gateway Controller (Softswitch):

© NOKIA 2017. ALL RIGHTS RESERVED.

Runs over any kind of access (such as Ethernet, ATM, SDH)

ALL call control in the media gateway controller

(MGC). RTP (Voice traffic)

 Both UDP and TCP are allowed as underlying

encoding. Session Description Protocol (SDP)

Session Description Protocol (SDP) RTP (Voice traffic)

SDP is used for negotiation between end points of

© NOKIA 2017. ALL RIGHTS RESERVED.

P-CSCF Media PSTN

User equipment: The user equipment is

© NOKIA 2017. ALL RIGHTS RESERVED.

Simplified view of the functional IMS architecture

© NOKIA 2017. ALL RIGHTS RESERVED.

Service H.248 AGCF

© NOKIA 2017. ALL RIGHTS RESERVED.

Voice Interoperability is a complex task as it requires

Interoperability is NOT the same as protocol