VoIP Over Wireless Network (06-88-563)Winter 2007

1. VoIP Over Wireless Network(06-88-563)Winter 2007 Course Advisor: Dr.K.Tepe Electrical & Computer Engineering University Of Windsor Presented by: Mohammad Izlal Haider (100650281)

2. Voice over IP - What is VoIP? • VoIP stands for Voice over Internet Protocol. • also referred to as IP Telephony. It is another way of making phone calls, though the ‘phone’ part is not always present anymore, as you can communicate without a telephone set. • VoIP is especially popular with long-distance calls.

3. The Wireless LAN and VoIP • A wireless LAN is one in which the devices and computers are connected without wires. • Instead of a hub, form which wires stem out to connect to the different machines in a wired network, you have a wireless router or hub, which may in turn be connected to an ATA(Analog Telephone Adapter).

4. Why a Wireless LAN? Mobility. This word itself says many things Just to give examples: • A medical team in a clinic needs to be able to communicate internally and externally while attending to emergencies, which implies being on the move. VoIP on a wireless LAN makes this possible for them, if each of them has a phone in their pocket. • A factory floor team, by nature very ‘bee-like’, will find it difficult to either remain glued to a fixed phone set or going to and from it for communication. Here again, VoIP service deployed on a wireless LAN within the company premises saves time, energy and nerves; and boosts productivity. • VoIP on Wi-Fi hotspots is a great thing for callers. Just like you take your laptop computer along with you for a business lunch or some revision among classmates, why not take your IP Phone or pocket PC along?

5. Reasons for Choosing Voice over IP - VoIP Advantages • Save a lot of money Studies have shown that, compared to using a PSTN line, using VoIP can potentially make you save up to 40 % on local calls, and up to 90 % on international calls. • More than two persons With VoIP, you can setup a conference with a whole team communicating in real time. • Cheap user hardware and software The only additional hardware you require besides your computer and Internet connection are a sound card, speakers and a microphone. • Abundant, Interesting and Useful Features Using VoIP also means benefiting from its abundant features which can make your VoIP experience very rich and sophisticated, both personally and for your business. You are thus better equipped for call management.Features also include Caller ID, Contact Lists, Voicemail, extra-virtual numbers etc Free! • The great thing about VoIP is that it taps additional value from the already existing infrastructure without additional costs. VoIP transmits the sounds you make over the standard Internet infrastructure, using the IP Protocol. This is how you can communicate without paying for more than your monthly Internet bill. • If you don’t use VoIP for voice communication, then you are most certainly using the good old phone line (PSTN– Packet-Switched Telephone Network). Contd..

6. VoIP Process

7. Making a VoIP Connection There are three ways in which you can make a VoIP connection • Computer to Computer This mode is the most common, as it is so easy and free. You need to have a computer connected to the Internet, with the necessary hardware to speak and listen (either a headset or speakers and a microphone). • Phone to Phone This mode is very handy, but is not as simple and cheap to set up as the other two. It implies using a phone set on each end to communicate. • Phone to Computer and vice-versa Now that you understand how you can use your computer, normal phones and IP phones to make VoIP calls, it is easy to figure out that you can call a person using a PSTN phone from your computer. You can also use your PSTN phone to call someone on his computer.

8. VoIP and Bandwidth - How Much Bandwidth Do I Need for VoIP? • For most people using the Internet as a communication medium, bandwidth happens to be the most expensive requirement, because it is recurring. For voice communication, the bandwidth requirements are more important, since voice is a type of data which is bulkier than conventional text. This implies that the greater the connection speed, the better the voice quality you can get. Today, broadband connection is common talk and getting cheaper and cheaper. • Broadband is unlimited connection (24 hours a day) at a speed much higher than that of dial-up’s 56 kbps. Most providers give at least 512 kbps today, which is largely sufficient for VoIP communication. This is the case for developed countries and regions. For other places, some users are still restricted to low connection speed at high prices

9. Common Bandwidths

10. "What is PSTN?" • Definition: PSTN stands for Public Switched Telephone Network. • It is the same thing as POTS (Plain Old Telephone System). • It is simply the worldwide telephone network. • It carries analog data. In constrast, VoIP uses digital data.

11. Circuit-switching • The important thing to look for in transmitting information over such a complex network is the path or circuit. The devices making up the path are called nodes. For instance, switches or routers are nodes. • In circuit-switching, this path is decided upon before the data transmission starts. The system decides on which route to follow, based on a resource-optimizing algorithm, and transmission goes according to the path. For the whole length of the communication session between the two communicating bodies, the route is dedicated and exclusive, and released only when the session terminates.

12. Packet-switching • The Internet Protocol(IP), just like many other protocols, breaks data into chunks and wraps the chunks into structures called packets. Each packet contains, along with the data load, information about the IP address of the source and the destination nodes, sequence numbers and some other control information. A packet can also be called a segment or datagram. • In packet-switching, the packets are sent towards the destination irrespective of each other. Each packet has to find its own route to the destination. There is no predetermined path; the decision as to which node to hop to in the next step is taken only when a node is reached. Each packet finds its way using the information it carries, such as the source and destination IP addresses.

13. Circuit Switching vs. Packet Switching • The old telephone system uses circuit switching to transmit voice data • VoIP uses packet-switching to do so. • The difference in the way these two types of switching work is the thing that made VoIP so different and successful.

14. Brief comparison • Circuit switching is old and expensive, and it is what PSTN uses. Packet switching is more modern. • PSTN call, you are actually renting the lines, with all it implies. See why international calls are expensive? With VoIP, you actually can use a network or circuit even if there are other people using it at the same time. There is no circuit dedication.

15. Encoding Voice • The process of transmitting a conversation via VoIP involves several important steps that are not present in a PSTN. The first is converting an analog voice waveform to a digital signal that may be transmitted by a data network.Several codecs exists for encoding voice signals for VoIP .

16. Voice compression • compressing CODECs are strongly recommended for saving bandwidth and increasing cell capacity. Using the basic non-compressing G.711 CODEC, which most VoIP gateways support, requires a bandwidth of 156 Kbps per direction at the basic rate (64 kbps raw data per call).

17. What is a Codec? codec is an algorithm , that is used to convert voice (in the case of VoIP) signals into digital data to be transmitted over the Internet or any network during a VoIP call. The word codec comes from the composed words coder-decoder or compressor-decompressor. Codecs normally achieve the following three tasks (very few do the last one): • Encoding – decoding • Compression – decompression • Encryption - Decryption • G.711 is the basic, most common CODEC is used for VoIP. • G.711 CODECs use Pulse Code Modulation (PCM) of voice frequencies at the rate of 64 Kbps.

18. Setting up and Maintaining Calls • In addition to delivering the actual voice content, another protocol is responsible for setting up, maintaining, and tearing down call sessions. • Session Initiation Protocol (SIP) and H.323 are two examples of protocols that perform this particular function. • Modeled after HTTP, SIP uses a reliable transport protocol to signal that one user wants to call another, a terminal is ringing,and a connection is established .

19. Session Initiation Protocol (SIP) SIP provides four basic functions. • SIP allows for the establishment of user location (i.e. translating from a user's name to their current network address). • SIP provides for feature negotiation so that all of the participants in a session can agree on the features to be supported among them. • SIP is a mechanism for call management - for example adding, dropping, or transferring participants. • Finally SIP allows for changing features of a session while it is in progress . • SIP is a catalytic protocol that delivers key signaling elements, which can turn a voice over IP network into a true IP communications network - a network capable of delivering next generation converged services. SIP is powerful, and yet simple.

20. SIP Methods

21. H.323 protocol • The main function of H.323 is to perform call control and management functions on a packet-switched network - it's really a session layer protocol. It's most often used (along with some other protocols) to provide VoIP services. • A router with H.323 configured will most often be acting as a "gateway". For instance, a router that has a connection to, and can translate communications to/from, a non-H.323 device. Typically a connection to a PBX, or to the PSTN. • It could be acting as a "gatekeeper". Gatekeepers are normally used for larger scale VoIP deployments and registers H.323 devices and gateways and helps to perform call setup and other functions. • Gateways and Gatekeepers are some of the components used to build a VoIP infrastructure, and routers are often used to provide these functions.

22. VoIP parameters • Voice Payload Size (Sample Size): The size in bytes of a packet generated by the CODEC. • Packet Size: The size in bytes of a packet, including RTP/UDP/IP overhead. • Packets Per Second (PPS): The number of packets generated per second. PPS=CODEC Bit Rate/Voice Payload Size. • Packet Duration (Inter Arrival Time or Sample Interval): The time between the start bits of two consecutive packets. Packet Duration=1/PPS.

23. MOS (Mean Opinion Score) • In voice communications, particularly Internet telephony, the mean opinion score (MOS) provides a numerical measure of the quality of human speech at the destination end of the circuit. • The scheme uses subjective tests (opinionated scores) that are mathematically averaged to obtain a quantitative indicator of the system performance. • A listener gives each sentence a rating as follows: (1) bad; (2) poor; (3) fair; (4) good; (5) excellent. MOS is the arithmetic mean of all the individual scores, and can range from 1 (worst) to 5 (best). • A MOS of 4.0 is typically the design target. However, in PSTN it may sometime get down to 3.7, and in cellular networks it may get as low as 3.2

24. QoS Metrics • Latency: Latency, or delay, is the time from when an audio signal is first recorded on one end of the communication to the moment it is heard on the opposite end. Often works analyzing QoS refer to this definition more specifically as absolute delay or mouth-to-ear delay . Evaluation studies show one-way delays surpassing 200 ms to be noticeable , and those over 400 ms to be unacceptable to many users . • Jitter: Jitter, the difference in the delay of successive packets , produces unnatural breaks in the voice when becoming too large. In practice applications reduce the effects of jitter by buffering packets • Packet Loss: Packet loss, especially prevalent in wireless networks, occasionally causes audio packets to not be available when it is their turn to be played. Since most audio is transmitted via an unreliable transport protocol such as UDP, there is usually some expected loss in congested paths or wireless links. • Echo:Though not directly measurable as a network statistic, echo is a noticeable effect exacerbated by network conditions, namely delay. Codecs usually contain at least some echo cancellation ability , making its effects less annoying. • Digitization Distortion: Like echo, the distortion due to the discretization of an analog signal is not directly measurable in the network, but may be perceivable by users. • Packet Size and Header Weight: A balance exists between sending many packets with a small delay and taking more time to assemble larger packets . The larger packets are more efficient because the headers that must be prefixed no matter what the size is.

25. Challenges/Solutions • Challanges : • low VoIP capacity in a WLAN and • unacceptable VoIP performance in the presence of coexisting traffic from other applications. • Voice quality • Bandwidth dependency • Power dependency • Emergency calls • Security • Solutions: • Voice multiplex–multicast M–M scheme can improve the VoIP capacity by close to 100%. The M-M scheme multiplexes the downlink VoIP packets into a larger multicast packet to reduce WLAN overheads. • Selecting The right VoIP GW will affect overall solution performances as cell capacity, latency and jitter.

26. Conclusion • By optimizing both the VoIPequipment and the wireless infrastructure, voice capacity can be increasedsignificantly. Making a difference that can absolutely improve the business model andturn the application to a very worthwhile revenue generator. • Moreover, by correctly configuring and building the wireless IP network, good telephony quality can be achieved, and overall voice and data capacity can be increased.

27. Reference • [1]Voice Over Internet Protocol (VoIP)- BurGoodie, proceedings of the IEEE, vol. 90, no. 9, september 2002 • [2] W. Jiang, H. Schulzrinne, “Comparison and Optimization of Packet Loss Repair Methods on VoIP Perceived Quality under Bursty Loss,” NOSSDAV, Miami, USA, pp. 73-81, May 2002. • [3] D. Chen, S. Garg, M. Kappes, and K. Trivedi, “Supporting VoIP traffic in IEEE 802.11 WLAN with enhanced medium access control (MAC) for quality of service,” Avaya Laboratories, Basking Ridge, NJ, Tech. Rep. ALR-2002-025, 2002. • [4]. Solutions to Performance Problems in VoIP Over a 802.11 Wireless LAN ,IEEE transactions on vehicular technology, vol. 54, no. 1, january 2005 • [5]. Voice Over IP ,Upkar Varshney, Andy Snow,Matt McGivern, and Christi Howard ,Communications of the ACM January 2002/Vol. 45, No. 1 • [6]. Establishing How Many VoIP Call a Wireless LAN Can Support Without Performance Degradation Ángel Cuevas Rumín ,Universidad Carlos III de Madrid Department of Telematic Engineering ,Ph.D Student Madrid, Spains , Eur Ing Chris Guy The University Of Reading School of System Engineering ,Head of School ,Reading, UK • [7] Anne Fladenmuller and Ranil De Silva, “The effect of Mobile IP handoffs on the performance of TCP,” Mobile Networks and Applications, Vol. 4, 1999, pp. 131-135. • [8] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol”,RFC 2543, Mar. 1999. • [9] Wanjiun Liao, “Mobile Internet telephony protocol: an application layer protocol for mobile Internet telephony services,” IEEE International Conference on Communications ICC '99, 1999 vol.1, pp: 339 –343. • [10] Mark A. Miller, Voice over IP: Strategies for the converged Network, M&T Books, 2000.