Transcript of "Analysis_of_VoIP_Telephone_Service.pdf"
Analysis of VoIP Telephone Service
Justin C. Zito
April 23, 2009
general internet or computer access. The cost of using VoIP is
Abstract—This paper takes a close look at the potential of VoIP less than the costs of a traditional telephone service, for both
telephone service, the benefits and challenges associated with local and long-distance services . Implementation and
VoIP and some concerning issues that arise as well. A brief study maintenance costs are greatly reduced, as much of the
is provided on the amount of packets both sent and received while
using VoIP telephone communication under simulated levels of
underlying infrastructure is already in place, since broadband
network congestion. Wireshark software is used to monitor the internet is the primary requirement. VoIP enables remote or
traffic generated during conversation and the X-Lite softphone rural businesses to better compete with larger, urban business
allows for computer interfacing. Packet data is collected, analyzed peers. The flexibility of VoIP allows service to users at any
and summarized over several protocols. The round trip time, location that provides broadband internet service. The
jitter, and ratio of packets transmitted and received are compared adapters used with analog phones are portable such that users
with the number of packets used for general internet applications.
A second study focuses on the effective data rates provided without a softphone or dedicated VoIP phone are still provided
between common codecs used in the transmission of the VoIP with service while traveling. The service allows most users to
audio stream. The packet size and packet transmission rates choose any area code as their „local‟ service area. Suitable for
govern the overall bit rate that is provided with each codec. those with family or friends living in a particular location, calls
between them are considered local and not charged as long-
Index Terms—Packet Monitoring, VoIP, Wireshark, VoIP distance. Some services provide extended areas of coverage
that may allow international calling as well. There are also a
bundle of free features that traditional phone services often
charge fees to use. Examples include voice mail, call
forwarding, call waiting, three-way calling, caller ID, call
V OIP, voice communications over IP networks such as
broadband internet, is becoming more widely used for
telephone service. The ever-growing uses of the internet are
block and call return applications.
resulting in vastly increasing numbers of users, data B. Challenges
transmissions, and of course, congestion. VoIP is essentially Analog phones require an adapter, which in turn requires
any voice communications transmitting through a broadband power. In the case of a power shortage, these adapters will not
internet service. work. Security issues such as phishing or spoofing scams may
VoIP providers often allow free calls to other VoIP not meet the high demands of classified data transmission,
subscribers, and usually charge a fee for calls to traditional making some wary to switching away from the already-proven
analog telephones. Services may provide calling to local, security of wired communications like POTS . Spammers
mobile, long-distance and international calls; anyone with a and hackers will most likely attack VoIP service as they do
phone number may receive VoIP calls, and the receiving user with virtually any internet service.
needs no extra equipment to talk to VoIP subscribers. There There is usually a noticeable delay across the channel; voice
quality using broadband transmission is not top quality, and
are three methods for making VoIP phone calls. Analog
any encryption methods implemented to enhance the security
telephones may be used with an external adaptor, a specific
will only increase latency and reduce the quality of voice.
VoIP phone is available with dedicated hardware, or a
There have also been some compatibility issues where VoIP
computer can be used to directly place calls with a softphone. protocols could not work with older firewalls of some LAN
The primary indication of the vast potential of VoIP for and WAN networks. Wireless or Wi-Fi areas for public use
massive numbers of users comes from both the cost benefits are not the most secure although there are more and more of
and the flexibility available with VoIP service. them arising at a relatively fast rate.
Often overlooked is the fact that almost all internet service
A. Benefits providers (ISPs) have a much smaller bandwidth for uploading
than downloading. Downloading files and streaming audio
VoIP service has potential for allowing combined voice and video is much more common than uploading data. As the
communications, email, fax, and even video conferencing. It number of VoIP users increases there may be negative effects
is capable of providing simultaneous phone service and in the quality due to the reduced upstream bandwidth. With a
general increase in VoIP users, there may arise bandwidth
problems for both directions of transmission when there is
excessive congestion within the network, resulting in buffer
overflow and packet losses.
Companies also face situations where they have already
invested in previous technologies. Keeping up with these may
be much less costly than implementing new technology. There
is also an issue with comfort levels; often times a company is
accustomed to their operations and feel wary to making radical
changes for several different reasons. Also, VoIP services are
not regulated in all countries, making regular development
Fig. 1. Wireshark network protocol analyzer; from left to right we can see the
packet number, arrival time, the source and destination IP addresses, the
C. Issues protocol used and the packet information.
The lack of regulation with internet-based phone service
will lead to other issues. For example, VoIP users cannot monitor the internet packet transmissions, and a softphone for
place direct 911 calls. Since VoIP service allows users to computer interfacing with the software. The service was
connect from any location that has broadband internet, it is provided through the gatorphone website which uses the
much more difficult for emergency operators to trace the Asterisk open source telephone engine.
location of calls. VoIP allows users to choose their area code Ten phone calls were used for data acquisition with each
for service, so caller ID is not a reliable method for locating call approximately five minutes in duration. The Wireshark
the caller either. The FCC made efforts by enforcing that network protocol analyzer captured the packet traffic
VoIP provider companies supply an Enhanced 911 (E911) generated while using VoIP. Fig. 1 shows a screen image of a
service . The rules associated with E911 assured that 911 live Wireshark capture. Two calls were monitored while there
calls are delivered to the user‟s local emergency operator.
was no internet browser opened, simulating a very low amount
Other rules were also enforced such as requiring all users to
of network traffic. No unnecessary applications were running
provide a call back number and location, and local phone
that communicated with the internet. Two calls were also
companies are required to share access to their E911 networks
with broadband telephone providers. monitored while an internet browser (Mozilla Firefox) was
The VoIP industry must (and indeed has) become concerned opened and connected. Six calls were captured in which the
with the possible effects that may result from these regulations. acquisition computer was either downloading data or
The power failure issue is still a concern as the E911 service streaming audio/video throughout the duration of the call, in
would not be of use in a power outage, where it may be needed attempt to simulate a high level of network congestion. Four of
most. Providing location information can be difficult in the calls were made while streaming and two while
emergency situations, although the FCC suggested tracking downloading. More data was collected for these situations
methods such as GPS. This however raised more concern with since they provided a more congested (and more realistic)
hardware issues. It would require more equipment costs which network simulation as opposed to the idle connections during
may be passed down as additional fees for the customers, the previous calls.
taking away from the initial benefits of using VoIP telephone
B. Data Collection
This paper aims to obtain some insight to the effect of VoIP A Wireshark capture was started prior to each phone call.
traffic generated during regular network congestion, and to The calls were made with one cellular phone and the X-Lite
provide an outlook on its potential for a massive number of softphone. After five minutes the calls are terminated and the
users. Calls are monitored while downloading, streaming Wireshark data saved for further analysis. Wireshark monitors
video, general browsing and with no additional programs the source and destination addresses of the packets as well as
opened. Standard codecs used with VoIP are compared as arrival time, the protocol and the codec used. It also provides
well. some information about the content of each packet (ACK,
Following are two different studies that look at the flow of NAK, lost packet, re-transmission, etc.).
VoIP data transmissions. The first experiment looks to gain The VoIP message data was transmitted using the Real-time
insight at the effects on the packet traffic in different levels of Transfer Protocol (RTP). In the beginning of each phone call
network congestion. The second study compares the data rates there are a few packets sent and received to establish the
of three common codecs used within VoIP services. connection and at the end to terminate the connection. These
packets use the Session Initiation Protocol (SIP). SIP packets
II. EXPERIMENT A are only used for invite requests, ringing status, and
termination requests. The analysis carried out in this paper
does not include the SIP data, as there are only ten or so
The experiment required a VoIP phone service, software to packets sent at the start and end of each phone call. Both the
RTP and SIP protocols run over the User Datagram Protocol connection and final termination for the phone calls (e.g., SIP
(UDP). packets). Available codecs range from speech-compression to
high quality audio codecs. The codec is necessary for secure
data transmission, and allows for a lower bit rate. It is used
either at the transmitter to encode or encrypt a signal or at the
receiving end to decode the message. A typical codec
employs an analog-to-digital converter (ADC) and an
encoding algorithm that is unique to each one.
In the second study, three common VoIP codecs are
compared among average throughput values. The codecs
analyzed were GSM, G.711 (µ-law) and iLBC. The GSM
(Global System for Mobile Communications) codec uses 8
kHz sampling rate and 13 kbps bit rate. The G.711 µ-law
encoding scheme uses companding to allow data to be sent at a
lower rate while preserving its quality. Companding uses
compression at transmission and expansion at the receiving
end, which provides higher resolution at lower voltages which
Fig. 2. Outgoing and incoming packets vs. round trip time; the red and blue corresponds with the range of most speech levels . µ-law,
describe the forward and reverse transmissions, respectively; the y-axis scales in oppose to A law, uses a different encoding algorithm, and µ-
from 0 to 50 msec. law is the American version whereas A law is used in
European countries. G.711 provides a 64 kbps bit rate and
The data used in the analysis accounts only for that also has 8 kHz sampling frequency (with 8 bits per sample). 8
transmitted during the phone call; it does not include the pre- kHz is common for speech sampling; human voice typically
or post-call data that was also saved with the Wireshark ranges from around 300 Hz to 4 kHz (rarely), so 8 kHz is
capture. Fig. 2 shows a plot of the packets round trip time for sufficient for most speech encoding. The iLBC (internet Low
both transmitted and received packets. In the Wireshark plot, Bit Rate Codec) allows for a varying data frame length, and
the forward direction corresponds with packets sent from the also uses 8 kHz sampling frequency. The bit rate for the iLBC
softphone (source), and the reverse direction denotes packets is at max 15.2 kbps, significantly lower than that of G.711 .
sent from the cellular phone to the softphone (destination). Three calls were captured for each of the codecs. Each call
Due to the large amount packets sent over five minutes, the lasted approximately two minutes. The average packet size,
figure shows about nine seconds worth of data, otherwise it average rate of individual packet transmission, and the average
bit rate were recorded for each call. The data collection
would look congested and unclear.
methods are the same as in the previous experiment (using the
Wireshark network analyzer and X-Lite softphone).
A. Experiment A
The transmissions of all outgoing packets were error-free,
with zero sequential errors and zero lost packets. The
incoming packets, as expected, did suffer some losses. For the
calls with no other internet communications, the RTP averaged
only 0.13% lost incoming packet. The situation was similar
Fig. 3. Outgoing and incoming packets vs. packet jitter; the black and green for the case where the internet browser was opened but there
lines describe the forward and reverse transmissions, respectively; the y-axis was no significant internet traffic. Here the average was only
scales from 0 to 25 msec.
0.01% dropped packets. The more interesting results came
Figure 3 displays a plot of the individual packet‟s jitter time. from the cases where the computer was loaded with several
The data lines in red and black correspond with the forward internet applications at once (i.e., high network congestion).
transmissions, and the blue and green data correspond with the Four calls were monitored while simultaneously streaming
reverse or incoming packets. video from a server; they averaged 0.13% packet loss with a
max recorded value of 0.18%. The downloading case was
very much like the streaming case. The averaged packet loss
III. EXPERIMENT B was 0.12% with a maximum of 0.14%.
VoIP systems use a session control protocol for connection Two major effects are quickly noticed during the VoIP calls
services and an audio codec for encoding or decoding of the while streaming or downloading. There is a significant drop in
audio stream. The control or service protocols provide initial the percentage of VoIP packets that are being processed as the
bandwidth must be shared with the other internet applications. packets for iLBC averaged 140 bytes per packet, comparable
Corresponding with the decrease in throughput of the VoIP to that of GSM (165 bytes/packet), though much less than then
packets, the download speeds were heavily bogged down the average size of G.711 packets (290 bytes/packet). iLBC
during the calls. carried less than half the amount of data per packet than the
leading G.711 codec. The reduction in both packet size and
Dir Max RTT RTP RTP Pkts Seq. % packet rate contributed to the drastically lower bit rate
(msec) Pkts Lost (%) Err. RTP associated with the iLBC codec. The average bit rate was 83
Kbps, supplying only 58% of the bit rate provided by the GSM
Stream_4 F 89.034 16708 0 (0) 0 43.57
R 458.045 16681 25 (0.15) 94
codec, and only 32% of the G.711 codec bit rate.
Stream_3 F 81.744 16577 0 (0) 0 57.32
R 289.018 16567 8 (0.05) 58 Avg pkts/sec Avg pkt size Avg bit rate
Stream_2 F 2075.467 15597 0 (0) 0 48.66 (bytes) (Mbps)
R 252.001 15583 28 (0.18) 18
Stream_1 F 86.436 15114 0 (0) 0 64.37 G.711 µ law 110 320 0.281
R 189.005 15096 18 (0.12) 23
Dwnld_2 F 83.337 15354 0 (0) 0 43.38 G.711 µ law 114 306 0.280
R 276.043 15329 22 (0.14) 22
Dwnld_1 F 88.407 15289 0 (0) 0 43.24 G.711 µ law 105 236 0.198
R 251.012 15272 15 (0.10) 15
Browse_2 F 92.326 15734 0 (0) 0 98.14 GSM 109 170 0.149
R 249.008 15731 1 (0.01) 1
Browse_1 F 861.232 14906 0 (0) 0 88.50 GSM 112 167 0.151
R 139.012 15080 0 (0) 0
None_2 F 80.377 12831 0 (0) 0 90.77 GSM 107 155 0.132
R 32.012 13192 8 (0.25) 1
None_1 F 981.696 14795 0 (0) 0 90.96 iLBC 69.5 118 0.065
R 148.223 15173 0 (0) 0
Table 1. Summary of data packets collected for ten VoIP calls. iLBC 67.7 146 0.079
Sequential errors in the packet delivery were also tracked. iLBC 85.2 154 0.105
They are negligible for the cases with no active internet or Table 2. Summary of data rates comparing three codecs.
when using a sole web browser, with one packet received out
of order in every 56,000 packets transmitted (18e-6 seq. error However, the difference in quality was not noticed by the
rate). The rate decreased during calls made while downloading users among the three codecs. There was a noticeable delay in
from the internet. Here the average was one sequence error for the channel during all of the phone calls, though the slight
every 1,660 packets (6e-4 seq. error rate). The calls made differences in latency were not user-distinguishable.
while processing streaming video recorded the largest
sequential error rate, averaging one error in every 660 packets
received (0.0015 seq. error rate). V. CONCLUSION
VoIP telephone service is growing in use daily. Businesses
B. Experiment B and household consumers are benefitting from the reduced
costs and flexibility of broadband telephone services. There
There showed a rather significant deviation between the
are several challenges associated with large-scale
average bit rates of the three codecs. The G.711 codec
implementation of VoIP services, although the potential
provided three to four times faster bit rate than the iLBC, and
benefits have companies investing much developmental
about twice that of the GSM codec. The packet rate was fairly
research. Issues and concerns arising have drawn attention
equal between the GSM and G.711 codecs. The improvement
from regulating agencies such as the FCC.
in bit rate came from the size difference of each packet for the
The priority of VoIP packets may become significantly
codecs. G.711 averaged 290 bytes per packet transmitted,
reduced when the network undergoes high congestion at peak
where the GSM packets average size was 165 bytes, about
service hours. This may become even more of an issue as the
57% of the size of the G.711 packets. The resulting bit rates
number of VoIP subscribers exponentially grows, and ISPs
directly reflected these differences. The average GSM codec
may have to allow for a more evenly distributed bandwidth
bit rate of 0.144 Mbps was precisely 57% of the bit rate for the
between uploading and downloading.
G.711 codec (0.253 Mbps). The iLBC suffered in both the
The first experimental study pointed out some potential
packet rate and packet size, reducing the average bit rate
problems that may require a bit more research before reaching
significantly. The averaged packet rate was only 74 packets
smooth VoIP service among large numbers of users. The calls
per second, much lower than the rate of the other two codecs
placed during low network traffic were close to error free, with
which averaged 110 packets per second. The size of the
less than 0.07% of packets lost. There were however a
noticeable number of lost packets when the network was more
congested, as simulated with the streaming and downloading
trials. Only 0.20% of these packets were lost, and this value
will probably grow larger when there are more users
contending for the available bandwidth. The priority of
packets was also reduced significantly. About 90% of the
traffic generated during low network congestion was RTP
packets associated with the VoIP service. This was reduced
below 50% as the network congestion was increased. This
trend will likely continue as the number of broadband
telephone subscribers increases.
The second study has shown some insight into the relative
efficiencies between commonly implemented VoIP codecs.
Among the three codecs analyzed, the average effective bit
rate varied from 83 kbps with iLBC to 0.253 Mbps with G.711
µ-law, which provided bit rates over three times as fast.
There is clearly a need for continuing codec research and
development, and the codec performance will likely become a
significant metric among competing VoIP providers.
This study was made with the gatorphone telephone service
supplied by Dr. H.A. Latchman. The X-Lite softphone and
Wireshark network protocol analyzer software allowed for the
data capture and statistical processing.
 “Benefits of VoIP.” VoIP Tutorial, 2005. Available: http://www.voip-
 “Security Issues in VoIP.” The New York Times Company, 2009.
Available: http://voip.about.com/od/security/Security_ Issues.htm.
 “Voice-Over-Internet Protocol.” FCC, March 2009. Available: http://
 EEL5718 Course Notes. H.A. Latchman, 2009.
 “Codecs.” VoIP Foro, 2009. Available: http://www.en.voipforo.com/
Justin C. Zito was born in Queens, NYC, in 1982 and moved to Orlando, FL
shortly thereafter. Received a BS in electrical engineering from the
University of Florida, Gainesville, FL, USA in 2006, and will receive a MSEE
in May 2009 from the same institution. Zito continues attendance at the
University of Florida as a doctoral candidate, with focus in MEMS actuators
and power electronics under the guidance of Dr. David P. Arnold.
He works in as a Research Assistant within an interdisciplinary
engineering lab at the University of Florida‟s Department of Electrical &
Computer Engineering. Recent projects have included the design
optimization of a zero-net mass-flux actuator, more commonly known as a
synthetic jet, and specifically on the electromagnetic actuation; and also on
the design of a power electronics system for plasma actuation used in active
flow control. He was published in a conference proceedings for the design of
an electrodynamically actuated microvalve for high-pressure flow using
COMSOL multiphysics and MATLAB software tools.
Mr. Zito is a member of the Interdisciplinary Microsystems Group (IMG)
at the University of Florida.