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  • 1. 1 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 [1]. 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 Codecs 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 I. INTRODUCTION 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 [2]. 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
  • 2. 2 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 somewhat problematic. 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 [3]. 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 service. 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 A. Setup 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
  • 3. 3 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 [4]. 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 [5]. 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). IV. RESULTS 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
  • 4. 4 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 Rcv‟d Pkts 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
  • 5. 5 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. ACKNOWLEDGMENT 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. REFERENCES [1] “Benefits of VoIP.” VoIP Tutorial, 2005. Available: http://www.voip- revolution-info-site.com. [2] “Security Issues in VoIP.” The New York Times Company, 2009. Available: http://voip.about.com/od/security/Security_ Issues.htm. [3] “Voice-Over-Internet Protocol.” FCC, March 2009. Available: http:// www.fcc.gov/voip/. [4] EEL5718 Course Notes. H.A. Latchman, 2009. [5] “Codecs.” VoIP Foro, 2009. Available: http://www.en.voipforo.com/ index.php. 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.