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Performance Analysis of Different Codecs in VoIP Using SIP

Performance Analysis of Different Codecs in VoIP Using SIP

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Ravi Shankar Ramakrishnan and P. Vinod kumar

International Institute of Information Technology

E- mail: 1 itsravishankar84@gmail.com, 2pvknet@gmail.com, 2vinodk_june07@net.isquareit.ac.in

ABSTRACT: Converged IP networks seek to incorporate voice, data, and video on the same infrastructure.
However, the integration of all types of traffic onto a single IP network has several advantages as well as
disadvantages. While reducing cost and increasing mobility and functionality, VoIP may lead to reliability concerns,
degraded voice quality, incompatibility, and end- user complaints due to changing network characteristics. The main
purpose of V oIP, Various CODECS used in V oIP and packet loss, Jitter, delay are analyzed and discussed.
KeywordsConverged IP Networks, Reliability Concerns, VoIP, CODECS.

INTRODUCTION

INTRODUCTION TO SIP

Session Initiation Protocol (SIP) is a peer-peer signaling

protocol for VoIP, developed by the IETF MMUSIC
Working Group and defined in RFC 2543. It is a proposed
standard for initiating, modifying, and terminating an
interactive user session that involves multimedia elements
such as video, voice, instant messaging, online games, and
virtual reality. SIP requires a simple core network with
intelligence embedded in endpoints; thus it is highly
scalable. It closely resembles HTTP and SMTP; thus SIP
sits comfortably alongside Internet applications.

oice over IP (VoIP) is the ability to transmit speech

over packet-switched IP networks. VoIP is an
acronym for Voice over Internet Protocol, or in more
common terms phone service over the Internet. If you have
a reasonable quality Internet connection you can get phone
service delivered through your Internet connection instead
of from your local phone company.
Placing a phone call using VoIP will create a digital
signal from the analog input, place those digital signals into
packets with source and destination network addresses and
finally send the information over the Internet or internal
company IP networks, thus bypassing the need for the
PSTN lines. However, it is important to note that both
source and destination terminals must support the particular
codec for proper encoding and decoding. Two popular
VoIP standards are H.323 and Session Initiation Protocol
(SIP).
Both standards allow for direct call establishment
between VoIP-capable terminals or the use of gatekeepers,
which can be used to negotiate connections between
endpoints. They can be used to translate between IP
addresses and telephone numbers, perform registration and
authentication functions, and manage bandwidth.

SIP-Allied Protocols
SIP interoperates with:
SDPTo describe the payload of message content and
characteristics
SAPfor advertising multimedia session via multicast
RSVPTo reserve network resources for providing
QoS
RTPFor real-time transmission
RTSPfor controlling delivery of streaming media
RADIUS For authentication
LDAPFor location discovery.

SIP Call Flow

Fig. 1: Basic VoIP Implementation

Fig. 2: SIP Call Flow Diagram

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Performance Analysis of Different Codecs in VoIP Using SIP

SIP Protocol Architecture

Codec Comparison
The following table lists the various codecs used in voice
over IP, and in particular SIP.
Codec
G.711
G.722

G.723.1
G.726

G.728
G.729
Fig. 3: SIP Protocol Architecture

REAL TIME PROTOCOL (RTP)

Audio, video, and multimedia services require the use of
RTP, which provides the necessary end-to-end delivery
requirements of time sensitive data. Both RTP and RTCP
were designed to run independently of the underlying
transport and network layers. RTP often runs in unison
with the User Datagram Protocol (UDP).

RTP packet format

A typical RTP packet includes a sequence number that
allows the receiver to reconstruct the data the sender has
sent in the appropriate order.

CODEC
A codec, which stands for coder-decoder, converts an
audio signal into compressed digital form for transmission
and then back into an uncompressed audio signal for
replay. It's the essence of Vo IP. It converts each tiny
sample into digitized data and compresses it for
transmission.

Common VoIP Codec

Codec
G.711
G.722
G.723.1
G.726
G.729

Comments
Delivers precise speech transmission.
Needs at least 128 kbps for two-way.
Adapts to varying compressions and
bandwidth is conserved with network
congestion.
High compression with high quality audio.
Lot of processor power.
An improved version of G.721 and G.723
(different from G.723.1)
Excellent
bandwidth
utilization.
Error
tolerant. License required.

Sampling
Rate (KHz)
8
16
16
16
8
8
8
8
8
8
unknown
8

Bandwidth
(Kbps)
64
48
56
64
5.3
6.3
16
24
32
40
16
8

License
Open Source
Open Source

Proprietary
Open Source

Open Source
Patented

QUALITY OF SERVICE (QOS)

In networking, quality can mean many things. In VoIP,
quality simply means being able to listen and speak in a
clear and continuous voice, without unwanted noise. QoS
(Quality of Service) is a major issue in Vo IP
implementations. The issue is how to guarantee that packet
traffic for a voice or other media connection will not be
delayed or dropped due interfer ence from other lower
priority traffic. Things to consider are Latency, Jitter and
Packet loss.

MEAN OPINION SCORE (MOS)

Quality Scale
Excellent
Good
Fair
Poor
Bad

Score
5
4
3
2
1

Listening effort Scale

No effort required
No appreciable effort required
Moderate effort required
Considerable effort required
No meaning understood with
reasonable effort

Coding techniques are such that speech quality degrades

as data rate reduces. However, the relationship is not linear.
Codec
G 711
G 726
G 726
G 728
G 729
GSM

Data Rate
64
32
63
16
8
13

Mos Score
4.3
4.0
3.8
3.9
4.0
3.7

RESULTS
The comparison between three different codecs has been
analyzed by implementing peer-t o-peer VoIP network
using SIP server and caller and callee.

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Mobile and Pervasive Computing (CoMPC2008)

SIP Server Active Calls

Fig. 4: SIP UAS Active Call timing Diagram

CALLEE: Delay Parameter

Fig. 7: Voice application traffic delay

CALLEE: Jitter Parameter

Fig. 5: SIP UAS Active Call Bar Diagram

CALLEE: Voice Traffic Received

Fig. 6: Voice application traffic received.

Fig. 8: Voice traffic delay variation

CALLER: Voice Traffic Received

Fig. 9: Voice application traffic received

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Performance Analysis of Different Codecs in VoIP Using SIP

CALLER: Delay Parameter

CONCLUSION
Thus we have described the various codecs in VoIP
implementation and analyzed three commonly used codecs
using peer-to-peer network scenario. These are common
narrow band codecs. It can be analyzed from the results
that G.711 is an ideal solution for PSTN networks with
PCM scheme. G.723 is used for voice and video
conferencing however provides lower voice quality. Music
or tones such as DTMF cannot be transmitted reliably with
G.723 codec. G.729 is mostly used in VoIP applications for
its low bandwidth requirement.

Fig. 10: Voice application traffic delay

CALLER: Jitter Parameter

REFERENCES
[1] Quality of Service for voice over IP- Cisco documentation.
[2] The effect of dynamic voice codec selection for active calls
on voice qualityThesis by Jered Daniel Ast.
[3] Packet Scan Users GuideGL Communication Inc.
[4] VoIPLecture notes by David Wang.
[5] Measurement Challenges for VoIP infrastructuresPrasad
Calyam, Internet2 VoIP Workshop.
[6] VoIP as a collaborative tool on client PCIntel White
paper.

Fig. 11: Voice traffic delay variation