Introduction to VoIP

Naveen Cheggoju§c

Adjunct Assistant Professor of Practice

Department of Electronics and Communication Engineering

VNIT Nagpur

ECL516 Converged Communication Networks

February 25, 2020

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 1/31
Contents

1 Introduction

2 VoIP challenges

3 VoIP codecs

4 VoIP voice quality

5 VoIP Protocol Stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 2/31
What is VoIP?

VoIP (Voice over Internet Protocol), sometimes referred to as Internet

telephony, is a method of digitizing voice, encapsulating the digitized
voice into packets and transmitting those packets over a packet
switched IP network by enabling people to use the Internet as the
transmission medium for voice communications.
This brings network converge into action, by combing voice and data
over samenetworks.
Necessity of VoIP:
1 Cost savings
2 Open standards
3 Multi-vendor interoperability
4 Integrated IP voice and data networks

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 3/31
PSTN and VoIP: Short summary

PSTN has made very impressive achievement in terms of coverage,

reliability, and ease of use.
The number of current telephones lines is estimated to be 1 billion.
The Internet does not offer the same degree of reliability as the
PSTN due to a variety of reasons like:
the complexity of multiple protocols
lack of standardization
multiplicity of equipment vendors and serviceproviders.
In addition, packet switched networks experience variable delays in
the transmissionprocess.
In contrast, the PSTN doesn’t suffer from variable delays, although it
can experience blocking when all the available circuits are being used by
other calls

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 4/31
PSTN vs VoIP

Table 1: PSTN vs VoIP

PSTN VoIP
Voice networks use circuit It uses packet switching
switching
Dedicated path between calling No dedicated path between
and Called party sender and receiver
Bandwidth is reserved in ad- It acquires and releases band-
vance width, as it is needed
Cost is based on distance and Cost is not dependent on time
time and distance

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 5/31
VoIP architecture I

Figure 1: Typical VoIP architecture

It can operate in 3 different scenarios:

1 Phone to phone communication
2 PC to phone communication PC
3 to PC communication

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 6/31
VoIP architecture II

These scenarios may remind us regarding a term called as

“convergence”, where data and voice are meeting on a single network to
form a converged network.
Though it has been introduced officially in 1975 and it became more
popular in 1995 with the introduction of new internet phone which
uses VoIP in the back-end.
By 1998, it has shown good improvement by occupying around 1% of
the world voice traffic.
It has become more and more popular by replacing the entire circuit
switching technology by its novel packet switched technology.
I think you are better enjoying the services of these improvements. . .
Let us now go in details of VoIP.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 7/31
Contents

1 Introduction

2 VoIP challenges

3 VoIP codecs

4 VoIP voice quality

5 VoIP Protocol Stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 8/31
VoIP Challenges I

Major challenges posed by VoIP networks are:

1 quality of service,
2 pricing and
3 security.

Quality of service
It is one of the most important concerns in voice communications,
determined by many factors, such as packet loss, speech coding options,
delay, echo andjitter.
The connection-oriented circuit-switched network provides each user with
dedicated bandwidth for the duration of each call, which results in extremely
low delay and jitter, and minimum disruption due to “noise”.
As we know, VoIP uses different codecs, and codecs affect the quality of
voice in a significant way.
So it is especially important to measure the quality of a voice call in a
standardized manner.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 9/31
VoIP Challenges II

Price
In a circuit switched network, an incomplete call is lost and does not
generate any revenue.
In the packet switched networks, there are no calls that are lost as such;
however, some of the packets may suffer delays above the acceptable
delay bound and are, similarly, not considered to constitute effective
throughput.
Just as a caller in a circuit switched network does not pay for an
incomplete call, the VoIP caller over a packet switched network in our
construct should not have to pay for packets that suffer an
unacceptable level of delay.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 10/31
VoIP Challenges III

Security
Voice over IP applications are generally designed to function over the
global Internet, although such solutions can be offered over private IP
networks, suchasenterprise networks.
Instances of violations of security over the Internet are common
occurrences that affect individuals, businesses as well as government
operations.
Voice over IP has not yet suffered many security violations, but the
potential for attacks on security is truly large.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 11/31
Contents

1 Introduction

2 VoIP challenges

3 VoIP codecs

4 VoIP voice quality

5 VoIP Protocol Stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 12/31
VoIP Codecs I

“CODEC” is a short notation for “coding and decoding”.

In VoIP different codecs are used to get the compressed form of voice
and reconstruct the same with high quality at the receiver.
Choosing a good codec is an important parameter in determining the
quality and efficiency of the call.
Quality will be more if the amount of compression is less, in other
words, more the bandwidth required more the quality of the voice.
This high bandwidth makes the call inefficient in terms of number of
calls handled in a given time.
Hence, one need to know regarding some important codecs used in
VoIP for picking the best suited codec for a given application.
Some of the most common and popular codecs in VoIP are:
G.711 G.723.1 G.726 G.729 GSM iLBC SILK

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 13/31
VoIP Codecs II

G.711:
One of the oldest and most popular codec of the VoIP.
Introduced in 1972, works very well at higher bandwidth.
It required 64kbps for one-way communication, which comes out to be
128kbps for two-way communication.
Delivers smooth and precise voice when network meets the bandwidth
specifications.
Uses PCM with in two modes, µ-law and A-law.
It has one of the best score of MOS (Mean opinion score), standing at
4.2.
G.723.1:
It was introduced around1990s.
Best thing to speak about this codec is that it works well in very low
bandwidth requirements.
It requires a bit rate of 5.3kbps or 6.3kbps.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 14/31
VoIP Codecs III

The excitation signal for the high rate coder is Multipulse Maximum
Likelihood Quantization (MP-MLQ) and for the low rate coder is
Algebraic-Code-Excited Linear- Prediction (ACELP).
It has MOS of 3.7 and 3.9 for lower and higher bit rates respectively.
G.726:
G.726 vocoder converts a 64 Kbps A-law or µ-law PCM channel to or
from a 40, 32, 24 or 16kbps channel.
G.726 can encode 13 or 14-bit PCM samples or 8-bit, A-law or µ-law
encoded data into 2, 3, 4, or 5-bit code words.
The algorithm encodes one sample at a time, the coding or decoding
delay is effectively zero, providing for robust, quality audio.
Allows several different bit rates, 40, 32, 24, and 16 kbps. Widely used is
32kbps.
It works well with packet to private branch exchange (PBX)
interconnections
G.729:
One of the better voice quality CODECs.
Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 15/31
VoIP Codecs IV

G.729 is an 8 kbps Conjugate-Structure Algebraic Code- Excited Linear

Prediction (CS-ACELP) speech compression algorithm approved by ITU-
T.
Requires very low processing power.
Making it more suitable for low end devices like mobile phones.
Special characteristics of this codec include error tolerance and
effective bandwidthutilization.
GSM:
High compression ratio.
Free and available in many hardware and software platforms.
The same encoding is used in GSM cellphones (improved versions are
often used nowadays).
It offers a MOS of 3.7, which is notbad.
iLBC: internet low bit-rate codec
Stands for Internet Low Bit Rate Codec.
It has now been acquired by Google and is free.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 16/31
VoIP Codecs V

Robust to packet loss, it is used by many VoIP apps especially those

with open source.
SILK:
SILK has been developed by Skype and is now licensed out, being
available as open-source freeware, which has made many other apps
and services to useit.
It is a base for the newest codec named Opus.
Best example is WhatsApp voice call.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 17/31
VoIP Codecs VI
Audio codecs
sampling frame
standardized bit rate
Name description rate si ze remarks
by (kb/s)
(kHz) (ms)
(ADPCM)
Intel, IMA ADPCM 32 8 sample
DVI
G.711 ITU-T Pulse code modulation (PCM) 64 8 sample mu-law (US, Japan) and A-law (Europe) companding
Adaptive differential pulse code
G.721 ITU-T 32 8 sample Now described in G.726; obsolete.
modulation (ADPCM)
Subband-codec that divides 16 kHz band into two subbands, each coded
G.722 ITU-T 7 kHz audio-coding within 64 kbit/s 64 16 sample
using ADPCM
Coding at 24 and 32 kbit/s for hands-free
G.722.1 ITU-T 24/32 16 20 See also
operation in systems with low frame loss
Extensions of Recommendation G.721
adaptive differential pulse code
Superceded by G.726; obsolete. This is a completely different codec than
G.723 ITU-T modulation to 24 and 40 kbit/s for digital 24/40 8 sample
circuit multiplication equipment G.723.1.
application
Part of H.324 video conferencing. D SP Group. It encodes speech or other
Dual rate speech coder for multimedia audio signals in frames using linear predictive analysis-by-synthesis coding.
G.723.1 ITU-T communications transmitting at 5.3 and 5.6/6.3 8 30 The excitation signal for the high rate coder is Multipulse Maximum
6.3 kbit/s Likelihood Quantization (MP-MLQ) and for the low rate coder is Algebraic-
Code-Excited Linear-Prediction (ACELP).
40, 32, 24, 16 kbit/s adaptive differential
G.726 ITU-T 16/24/32/40 8 sample ADPCM; replaces G.721 and G.723.
pulse code modulation (ADPCM)
5-, 4-, 3- and 2-bit/sampleembedded
G.727 ITU-T adaptive differential pulse code var. ? sample ADPCM. Related to G.726.
modulation (ADPCM)
Coding of speech at 16 kbit/s using low-
G.728 ITU-T 16 8 CELP. Annex J offers variable-bit rate operation forDCME.
delay code excited linear prediction
Coding of speech at 8 kbit/s using
G.729 ITU-T conjugate-structure algebraic-code-excited 8 8 10 Low delay (15 ms)
linear-prediction (CS-ACELP)
G Regular致ulse Excitation Long茅erm
ETSI 13 8 22.5 Used for GSM cellular telephony.
SM 0 Predictor (RPE-LTP)
6.10
LPC10e
(FIPS US Govt. Linear-predictive codec 2.4 8 22.5 10 coefficients.
1015)

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 18/31
Contents

1 Introduction

2 VoIP challenges

3 VoIP codecs

4 VoIP voice quality

5 VoIP Protocol Stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 19/31
Quality of speech in VoIP

Different codecs we had seen until now compress and decompress the
speech with highest quality possible.
But this is not the end, for perceiving the high quality voice.
There are some others parameters that can effect the voice quality at
the receiver.
Sometimes it may happen even with the best codecs voice quality
may be poor.
The parameters that governs this loss in voice quality are:
delay,
jitter, and
packet loss.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 20/31
Delay I

Due to the interactive nature of voice communication, delay becomes a

primary parameter of concern in the QoS (Quality of Service) measure
for VoIP networks.
This delay may occur due to different parameters:
1 transmission delay,
2 queuing delay,
3 processing delay and
4 propagation delay.
Transmission delay: Delay due to the variation in the required
bandwidth and the available bandwidth of the channel. It is
dependent on the channel capacity in bits per second (bps).
Queuing delay: It is the time the packets are queued in the buffer
before being processed.
Processing delay: It is incurred at the end points, e.g., in processing
packet headers, and in coding/decoding voice signals.
Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 21/31
Delay II

Propogation delay: It depends on the distance traveled and the

transmission medium, such as coax, fiber, or wireless channel

Table 2: Delay specifications for voice

Delay Impact
Below 150ms Acceptable for most userapplications
150 - 400ms Acceptable for international calls
Above 400ms Unacceptable for general network planning pur-
poses, especially in the case of transporting voice
in packet switched networks

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 22/31
Jitter

Jitter is delay variation.

It can lead to the gaps in the play out of the voice stream.
The jitter can be compensated by maintaining a play out buffer on the
receiver side, which processes the incoming packets in such a way that
packets arriving earlier than average time are buffered for a longer
period than those arriving later.
This means that the received voice stream can be recovered at a
steady rate.
In addition, arriving voice packets that exceed the maximum length of
the jitter buffer are discarded.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 23/31
Packet Loss

Packet loss in a network can occur in many different ways.

The most discussed reason is network congestion.
Packets may get lost in the high traffic networks, or, may get delayed
due to the traffic.
This delay, if more than the average delay provided by the receiver,
creates the packet loss.
Jitter can also be one of the reason for the packet loss, because it is
caused by the non-uniform arrival of the packets at the receiver.
Packet loss introduces impairment in the successive speech and
creates disturbances in thevoice.

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 24/31
Contents

1 Introduction

2 VoIP challenges

3 VoIP codecs

4 VoIP voice quality

5 VoIP Protocol Stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 25/31
VoIP protocol stack I

Figure 2: OSI vs VoIP protocol stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 26/31
VoIP protocol stack II

Figure 3: TCP/IP vs VoIP protocol stack

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 27/31
VoIP protocol stack III

Figure 4: VoIP protocol stack with codecs

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 28/31
VoIP protocol stack IV

Figure 5: VoIPencapsulation

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 29/31
VoIP protocol stack V

Next chapter discusses about H.323 (Standards of equipment),

MEGACO (Gateway control protocol, H,248) and SIP (Session
Initiation Protocol).

Naveen Cheggoju§c (VNIT) ECL516 Converged Communication Networks February 25, 2020 30/31