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An Analysis of VoIP Communication and Overview of Bangladesh Practical

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Article in International Journal of Computer Theory and Engineering · January 2013

DOI: 10.7763/IJCTE.2013.V5.752

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 International Journal of Computer Theory and Engineering, Vol. 5, No. 3, June 2013

An Analysis of VoIP Communication and Overview of

Bangladesh Practical Field
N. A. Shafi, Member, IACSIT, O. Farrok, and M. M. Ali


compressed and then encoded into digital voice streams by
Abstract—In this paper the latest development G.729 which the codec [4]. The VoIP protocol stack is illustrated in Fig. 1.
is an audio data compression algorithm has been analyzed and Voice packets are transmitted over the IP network, and the
implemented to save a noticeable bandwidth. Different effects reverse processes of decoding and depacketizing are
of AnnexB=yes and AnnexB=no has been observed in the G.729
codec which has the standard bandwidth of 8 kbps. As most of
accomplished at the receiver. A playout buffer is used by the
the internet subscribers of Asian countries use very low internet receiver to smoothen the speech by getting rid of delay jitter.
bandwidth the goal of this analysis is to propose to change some Packets arriving later than the playout time will simply be
parameters of this system so that the quality of voice may be discarded. Some other components such as voice/silence
kept in a tolerable limit using only 5kbps to 6kbps where this detector, loss/error concealment and echo canceller, are also
codec uses at least 6.4kbps. In a real life environment it is tested included in the system to enhance the functionality and
practically that it is possible to transmit voice satisfactorily
using less than 6.4 kbps by changing some parameter described
performance of VoIP systems. The major metric to evaluate
in this paper. the user-perceived voice quality is the Mean Opinion Score
(MOS).
Index Terms—G.729 codec, AnnexB=yes, AnnexB=no, RTP
(real-time transport protocol), VAD (voice activity detector). Application Layer Voice
Transport Layer RTP RTCP SIP H.323
UDP
I. INTRODUCTION Network Layer IP
Link Layer NIC
The G.729 codec perform voice compression at bit rates Physical Layer Ethernet
that vary between 6.4 and 12.4kbps [1]. It is an audio data
Fig. 1. VoIP protocol stack.
compression algorithm for voice that compresses digital
voice in packets of 10 milliseconds duration. It is officially
Voice codecs are standardized by the International
described as Coding of speech at 8 kbit/s using
Telecommunication Union-Telecommunication (ITU-T),
conjugate-structure algebraic code-excited linear prediction
such as G.729 with 8 kbps, G.723.1a with 5.3/6.3 kbps, etc.
(CS-ACELP). Because of its low bandwidth requirements,
The output voice stream then enters the packetizer to
G.729 is mostly used in Voice over Internet Protocol (VoIP)
generate constant bit rate (CBR) audio packets with RTP
applications where bandwidth must be conserved. Standard
(RTCP)/UDP/IP header where RTP and RTCP [5] are
G.729 operates at a bit rate of 8 kbit/s, but there are
Real-time Transport Protocol and Real-Time Control
extensions, which provide rates of 6.4 kbit/s (Annex D, F, H,
Protocol, respectively, which are designed to support
I, C+) and 11.8 kbit/s (Annex E, G, H, I, C+) for worse and
real-time multimedia applications with stringent delay
better speech quality, respectively.
constraint over unreliable User Datagram Protocol (UDP).
To achieve good quality low-bit-rate silence compression,
Besides these, call set-up signaling protocols, such as Session
a robust frame-based voice activity detector module is
Initiation Protocol (SIP), are used for establishing VoIP
essential to detect inactive voice frames, also called silence or
connections. SIP, defined in RFC 2543 of Internet
background noise frames. The achievement of bit-rate
engineering task force (IETF), is a signaling protocol for
savings for coded speech at average rates as low as 4 kb/s
Internet applications, e.g. conferencing, telephony, events
during normal speech conversation while maintaining
notification, and instant messaging.
reproduction quality [2]. G.729 Codec is closely relation to
RTP [3].The packet-level VoIP system performance has been
focused. The analogue voice signals are digitized,
II. DESCRIPTION OF THE SYSTEM
G.729 has been extended in Annex B (G.729b) which
provides a silence compression method that enables a voice
Manuscript received November 8, 2012; revised December 18, 2012. This
work was arranged and supported by Zamir Telecom Limited, Authors would activity detection (VAD) module. It is used to detect voice
like to express their deepest sense for the technical and financial support. activity in the signal. It also includes a discontinuous
N. A. Shafi is with the Zamir Telecom Limited, 4-6 Davenant Street, Unit transmission (DTX) module which decides on updating the
# A, London E1 5NB, England (e-mail: nahid_apee@yahoo.com).
O. Farrok is with the Department of Electrical and Electronic Engineering, background noise parameters for non speech (noisy frames).
Ahsanullah University of Science and Technology (AUST), Dhaka-1208, It uses 2-byte Silence Insertion Descriptor (SID) frames
Bangladesh (e-mail: omarruet@gmail.com). transmitted to initiate comfort noise generation (CNG). If
M. M. Ali is with the Department of Electrical and Electronic Engineering,
Rajshahi University of Engineering and Technology (RUET), Rajshahi-6204, transmission is stopped, and the link goes quiet because of no
Bangladesh (e-mail: mmali.ruet@gmail.com). speech, the receiving side might assume that the link has been

DOI: 10.7763/IJCTE.2013.V5.752 574

 International Journal of Computer Theory and Engineering, Vol. 5, No. 3, June 2013

cut. By inserting comfort noise, analog hiss is simulated

digitally during silence to assure the receiver that the link is
active and operational [6].
A Voice Activity Detector (VAD) with a comfort noise
generator (CNG), achieves silence compression, which is
very important in modern telecommunication systems [7]. In
multimedia communications a VAD guarantees simultaneous
voice and data applications; in Universal Mobile
Telecommunication Systems (UMTS), it reduces the average
bit rate; finally, in a cellular radio system using the
Discontinuous Transmission (DTX) mode, it reduces
co-channel interference and power consumption in portable
equipment. The paper presents a performance evaluation and
comparison of recent ITU-T and ETSI voice activity
detection algorithms. The last ITU-T VAD standard is Rec.
G.729 Annex B [8], developed for fixed telephony and Fig. 4. 1st portion of call trace report for receiver end of VOS switch as
multimedia communications [9]. The G.729 Codec receives AnnexB= yes.
and transmits audio packet two ways in the VOS switch.
AnnexB=yes is explained in Fig. 2.
SPEECH
SPEECH
SPEECH

SPEECH
SPEECH
SPEECH
SP
SP

Fig. 2. Block diagram of AnnexB=yes.

There is 8kbps packet. Two packets is silence compression

(SP) that means the gap of two speeches. The silence
compression generates the room noise. AnnexB=no is
explained in Fig. 3.
SPEECH

SPEECH

SPEECH

SPEECH

SPEECH

SPEECH

Fig. 3. Block diagram of AnnexB=no.

There is 6kbps packet. Two packets do not generate

silence compression. Recently, Payload characteristic
research (G.723 Annex A and G.729 Annex B) is running in Fig. 5. 2nd portion call trace report for receiver end of VOS switch as
AnnexB= yes.
the world [10].

III. DISCUSSION AND RESULT

G.729 has an optional Annex B [11], which specifies the
use of silence suppression and comfort noise generation. In
typical speech, only one person talks at a time. Therefore,
speech consists of periods of talking (called talkspurts),
followed by periods of silence. Additional compression can
be achieved by discovering the silence periods. Older
approaches would send either nothing for the silence periods,
or would send a simple energy value, which the decoder
would use to insert white noise. The practical example of the
call traces are shown in Fig. 4. to Fig. 9. However, in
environments with loud and non-stationary background noise,
both approaches are inadequate. The algorithm operates by
first making a Voice Activity Detection (VAD) decision in
Fig. 6. 1st portion of call trace report for receiver end of VOS switch as
each frame. AnnexB= no.

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 International Journal of Computer Theory and Engineering, Vol. 5, No. 3, June 2013

(6.4-5/6.4)×100% = 21.875%. So average bandwidth saving

is (6.25 + 21.875)/2 = 14.0625% or approximately 14%.

Fig. 7. 2nd portion of call trace report for receiver end of VOS switch as
AnnexB= no. Fig. 9. 2nd portion of call trace report for transmitter end of VOS switch as
AnnexB= no.

IV. CONCLUSION
G.729 has been extended with various features, commonly
designated as G.729a and G.729b. Dual-tone
multi-frequency signaling (DTMF), fax transmissions, and
high-quality audio cannot be transported reliably with this
codec. DTMF requires the use of the RTP Payload for DTMF
Digits, telephony tones, and telephony signals as specified in
RFC 2833. Research for this article was done by means of a
literature study and practical work knowledge in Bangladesh.
The literature review aims to provide business management
with a review of the development of VoIP in Asian countries.
The literature review included a study of published and
internet articles, books and literature on the management of
information systems [13]. The research department of VOIP
update is G.729 codec AnnexB=no if the audio packet size is
less than 6.4kbps then a good voice quality is achieved in a
low band width of 5kbps to 6kbps only that saves average
bandwidth of approximately 14%.

Fig. 8. 1st portion of call trace report for transmitter end of VOS switch as APPENDIX
AnnexB= no. VOS switch, VPS switch, various dialer, SQL database,
So, noise will be generated which is undesired [12]. When firewall system, STM device and CISCO router & switch,
we are decreasing RTP packet size for low network area, it is OS-Windows server and Linux server etc.
not possible to achieve good quality for audio codec G.729. If
it is possible to decrease frame size audio codec G.729, then ACKNOWLEDGMENT
we may get it good voice quality. We are still not getting This research work is inspired and supported by Zamir
good voice quality at low network as AnnexB=no. But Telecom Limited, Situated in 4-6 Davenant Street, Unit # A,
AnnexB=no is better than AnnexB=yes. If the audio packet London E1 5NB.
size is less than 6.4kbps as AnnexB=no, that is possible for
good voice quality at low band width area. It is also possible REFERENCES
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 International Journal of Computer Theory and Engineering, Vol. 5, No. 3, June 2013

[2] ITU-T Recommendation, “G.729 Annex B: a silence compression N. A. Shafi was born in Rajshahi, Bangladesh on July
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