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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Advanced Networking
Multimedia networks-Part 2- VoIP

Fayssal BENDAOUD

f.bendaoud@esi-sba.dz

Options: SIW, IASD


November 11, 2024

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Outline

Introduction

VoIP

IMS: IP Multimedia Subsystem

Audio coding

VoIP protocols
H.323
SIP Session Initiation Protocol

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Introduction

• The traditional method used for transmitting voice is


PSTN.(public switched telephone network)
• Now more and more communication is done in digital format and
transported via data networks such as internet.
• As data traffic is much faster than telephone traffic, so we prefer
to send voice over data networks.

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Circuit switched

• Circuit Switching: A dedicated communication path is


established between devices for the entire interaction.
• Exclusive Channel: Once connected, a channel is reserved solely
for the communicating parties.
• Predictable Flow: Ensures continuous, consistent data flow
without interruptions.
• Data Transmission Approach: Provides a reliable and
uninterrupted communication line during the session.

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Packet switched

• Packet Switching: Data is split into packets routed individually


across the network.
• Efficiency: Allows multiple users to share network resources,
optimizing bandwidth.
• Flexibility: Packets take the most efficient path, adapting to
network conditions.
• Variable Delays: May introduce delays, as packets can take
different routes to the destination.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

VoIP

• VoIP, or ”Voice over Internet Protocol,” allows voice calls to be


made over the Internet instead of traditional phone lines.
• This enables calls to be made and received on various devices,
including laptops, tablets, smartphones, and VoIP-compatible
office phones.
• VoIP is cost-effective for both local and international calls, with
many providers offering unlimited minutes.
• Additionally, it includes advanced features like call waiting, call
routing, auto-attendants, and multiple phone numbers at no extra
cost.

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Advantages of VoIP

• Cost-effective: VoIP offers lower costs for domestic and


international calls by using the internet, eliminating
long-distance charges, which is beneficial for businesses with
global clients or remote teams.
• Flexibility: VoIP enables calls on any internet-connected device,
allowing employees to stay connected remotely. This flexibility
supports remote work and enhances productivity.
• Rich features: VoIP systems offer advanced features like call
forwarding, voicemail-to-email, video conferencing, virtual
numbers, and auto-attendant menus, which improve
communication and collaboration in organizations

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Advantages of VoIP

• Scalability: VoIP systems are highly scalable, allowing


businesses to add or remove users easily without physical
changes, making them ideal for growing or fluctuating
communication needs.
• Reliability and redundancy: VoIP systems offer redundancy
and fail-over features for high reliability, re-routing calls to
backups or mobile devices during outages to minimize business
disruptions.

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How does VoIP work

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

How does VoIP work

• Instead of using traditional phone lines, VoIP converts your voice


into digital data.
• This data is then transmitted over the Internet to the person we
are calling.
• At the receiving end, the digital data is converted back into
audio, allowing the other person to hear your voice.

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How does VoIP work

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Devices used in VoIP

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Soft-phones

• A soft-phone is software that enables internet calls via VoIP on


any connected device, like a laptop, smart-phone, or tablet.
• Popular soft-phone apps, such as Skype, Zoom, WhatsApp, and
Viber, allow users to make calls, chat, and hold video
conferences from anywhere in the world.

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Soft-phones

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Challenges of VoIP
• Interoperability issues across networks and providers:
Different communication networks and service providers might
use different technologies or standards, making it challenging for
users on one network to seamlessly communicate with users on
another.
• Inefficient scaling to meet the rising demand for VoIP and
multimedia:
As the number of users and the demand for services increase, the
network infrastructure may struggle to scale efficiently to handle
the growing traffic.
• Mobility issues across devices and networks:
Users may face difficulties maintaining communication services
seamlessly when switching between different devices or moving
across various networks.
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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Challenges of VoIP
• Quality of Service concerns, particularly in real-time
communication:
Real-time communication services, like voice and video calls,
require consistent and high-quality performance. Issues such as
delays or poor audio/video quality can be detrimental.
• Security vulnerabilities in VoIP over IP networks:
VoIP calls and data transmitted over IP networks can be
susceptible to security threats like eavesdropping, unauthorized
access, or malicious attacks.
• Limited flexibility for introducing innovative services:
The existing infrastructure may not easily support the
introduction of new and innovative communication services as
technology evolves.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

IMS

• IMS (IP Multimedia Subsystem) is an architectural framework


that enables multimedia services over IP networks, supporting
the convergence of voice, video, and data on a single all-IP
infrastructure.
• IMS was developed to address the challenges associated with the
integration of traditional circuit-switched telecommunication
networks and emerging IP-based networks
• IMS offers a standardized platform for multimedia services,
enabling operators to deliver VoIP, video calls, messaging, and
other real-time applications over a common IP infrastructure.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

IMS standardization for mobile and fixed-line networks

• The IP Multimedia Subsystem (IMS), first specified in 2003 by


3GPP, was originally designed for 3G UMTS networks and is
now also used in 4G LTE. It defines how multimedia connections
and services are managed via the IMS core network, with
services accessible through open interfaces. Although initially
for mobile networks, IMS was later extended to include
fixed-line networks.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

VoIP, IMS & Mobile Network Generations

• 2G & 3G
In the early generations of mobile networks (2G and 3G), voice
services primarily relied on circuit-switched networks. VoIP was
not the primary method for voice communication during this
period.
• 4G LTE:
• With the advent of 4G/LTE, there was a shift toward an all-IP
network architecture. IMS became more significant as it allowed
the integration of VoIP services over LTE networks.
• VoLTE (Voice over LTE) is a specific implementation of VoIP
within the IMS framework, enabling high-quality voice calls over
LTE networks.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

VoIP, IMS & Mobile Network Generations


• 5G:
• IMS continues to be a crucial component in 5G networks,
ensuring the support for a wide range of multimedia services.
• Vo5G (Voice over 5G) is an evolution of VoLTE, and it leverages
IMS to provide voice services over the latest 5G networks

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VoIP Variants

• IMS+LTE=VoLTE
• IMS +5G=VoNR
• IMS+WiFi=VoWIFI

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Audio coding

• VoIP means we are using packet mode


• Voice digitization.
• Digitised data may be large
• Compress to reduce size.

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The general scheme of VoIP

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

The general scheme of VoIP

• Sampling is the process of periodic capturing and recording of


voice. The result of sampling is called a pulse amplitude
modulation (PAM) signal.
• Quantization is the process of assigning numeric values to the
amplitude (voltage) of each of the samples on the PAM signal
using a scaling methodology.
• Encoding is the process of representing the quantization result
for each PAM sample in binary format.

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Sampling

• To discretise the signals, the gap between the samples should be


fixed. That gap can be termed as the sampling period Ts.
• fs = t1 .
s
• fs : sampling rate or frequency, ts : sampling time or period
• To reconstruct an analog signal from a digitised one, the
sampling frequency should be such that the data in the message
signal should neither be lost nor it should get overlapped. Hence,
a rate was fixed for this, called as Nyquist rate.
• Suppose that a signal with W is the highest frequency. For
effective reproduction of the original signal, the sampling rate
should be twice the highest frequencyfS = 2 ∗ W . The frequency
rate becomes Nyquist rate

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Sampling

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Sampling

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Quantisation

• Quantization is the process of rounding of the sample value to


the nearest quantization level. Remember that number of
quantization levels is predefined.
• If n is number of bits used to represent the sample. Then, q = 2n
is number of quantization levels

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Pulse Code Modulation (PCM)

• One common method of converting analog voice signal to digital


voice signal is pulse code modulation (PCM), which is based on
taking 8000 samples per second and encoding each sample with
an 8-bit binary number.
• PCM, therefore, generates 64,000 bits per second (64 Kbps); it
does not perform compression.
• Each basic digital channel that is dedicated to transmitting a
voice call within PSTN (Digital Signal 0= DS0) has a 64-kbps
capacity, which is ideal for transmitting a PCM signal.

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Codecs

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Remark

• The human speaking voice typically has frequencies of up to


4000 Hz, so a sampling rate of 8000 Hz is sufficient to sample it
with an acceptable quality. This is why a sampling frequency of
8 kHz is often found within codecs used for VoIP
• MOS is an ITU standard method of measuring voice quality
based on the judgement of several participants; therefore, it is a
subjective method.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

VoIP protocols

• International Telecommunications Union (ITU)


• H.323 – Visual Telephone Systems and Equipment for Local Area
Networks which Provide a Non-Guaranteed Quality of Service
• Internet Engineering Task Force (IETF)
• Session Initiation Protocol (SIP)
• Media Gateway Control (Megaco)
• Signal Transport (SigTran)

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Outline

Introduction

VoIP

IMS: IP Multimedia Subsystem

Audio coding

VoIP protocols
H.323
SIP Session Initiation Protocol

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

H.323

• Standard covering multimedia communications over LANs that


do not provide a guaranteed Quality of Service
• Entities
• Terminals
• Gateways
• Gatekeepers
• MCUs
• Protocols
• Parts of H.225.0 - RAS, Q.931
• H.245
• RTP/RTCP
• Audio/video codecs

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H.323

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H.323: components

• The terminals: Participants in a multimedia session


• Gateways:
• Ensures interconnection between the H.323 network and other
telephone networks (PSTN, SIP, etc.)
• Gatekeepers:
• Responsible for customer registration and authentication
• Takes care of address translations (phone number/IP address)
• Multipoint control units MCUs: Allows clients to connect to
data conferencing sessions

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H.323 protocol stack

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H.323 protocol stack

• Signalling:
• RAS (H.225): Registration Admission Status (on UDP) for
registration and authentication
• Q.931 (over TCP) for call initialization and control
• Negotiation:
• H.245 (over TCP) for codec negotiation and media stream
management
• Transport of information:
• RTP for transferring voice, video and digitized data via codecs
• RTCP for supervision

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

Outline

Introduction

VoIP

IMS: IP Multimedia Subsystem

Audio coding

VoIP protocols
H.323
SIP Session Initiation Protocol

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

SIP Session Initiation Protocol

• SIP is an application layer control (signaling) protocol for


creating, modifying, and terminating sessions with one or more
participants.
• Security aspect
• End-to-end encryption
• Hop by hop encryption
• Hop-by-hop encryption of the ”Via” header field to hide the route
the request took.
• Microsoft migrated from net-meeting (H.323) to MSN
messenger / windows XP(SIP)
• SIP message format is based on the HTTP message format,
which uses a human readable, text-based encoding

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

SIP Session Initiation Protocol

• Manage the phases of a phone call


• Location of a terminal (address)
• Negotiation of media type and communication parameters (voice
or video, codecs, etc.)
• Call establishment and follow-up
• Management of advanced features such as encryption, errors, etc.

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

SIP Session Initiation Protocol

• A session is established with requests and responses like as HTTP


• Queries are defined through six methods: REGISTER, INVITE,
ACK, CANCEL, BYE, OPTIONS.
• It uses of SDP (Session Description Protocol) for media content
(audio, video) and transport (RTP)

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

SIP components

• User Agent: User of the SIP network


• Gatekeeper: Manages user information (addresses) for a
network or network segment
• Proxy: Receives, processes and, if necessary, forwards SIP
signaling messages

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Introduction VoIP IMS: IP Multimedia Subsystem Audio coding VoIP protocols

SIP components

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