Managing Business Activities and Using Internet Telephony and VoIP - Assignment Example

Network Question Network Questions Affiliation Part VOIP Quality of Service (QoS In your own terms define why QoS is an important part of VOIP. The QoS is a significant element of VoIP, in other words we can say that it is a most important part of the VoIP structure. The main consideration of this factor is for the businesses that are moving to internet telephony. For the reason that majority of the businesses now use internet telephony and VoIP, so the QoS is mainly necessary for them to effectively manage business activities. However, the businesses that switch to a low-cost VoIP service are frequently surprised to find out that they have come into a world where service quality can be intangible and hard to attain. Since, there are a number of limitations, such as latency (a holdup in data packet delivery), network jitter (data receiving out of order), packet loss (missing data) as well as there are numerous other factors that can degrade a voice connections quality, that take towards the disconnection, distorted audio and even total VoIP system failure. In this scenario, a network with a quality of service has the capability to transfer data traffic through a very small amount of delay in a situation in which a lot of users share the similar network. Moreover, in this scenario, the low-priority traffic fails and high priority traffic offers most excellent services. On the other hand, if the network is not capable to have similar bandwidth that can support even high-priority traffic might not get through. Furthermore, the traffic engineering, that facilitates QoS, is trying to make it certain that the network can be able to offer the expected traffic loads (Edwards, 2007) and. 2. Describe the key features and differences between Int-Serv and Diff-Serv VOIP QoS services. Integrated Services (Int-Serv) is an architecture that provides QoS on the internet as well as intranets. The importance of IntServ designers was to set aside a number of pieces of network bandwidth for traffic like that real-time voice as well as video that need low jitter, low delay as well as guaranteed bandwidth. Moreover, the IntServ working group has developed resource reservation protocol (RSVP), which is a signaling method to specify QoS needs in the course of a network (Terrasa, Saez, & Vila, 2003). On the other hand the main aim of developing differentiated services architecture is to offer a way of offering a range of services on the internet without the necessity of per-flow state as well as signaling in each router. In addition, the DiffServ follows the standard of traffic classification, where every data packet is divided into a few number of traffic classes, rather than distinguishing network traffic foundational on the needs of an individual flow. Thus, it outlines that per flow resources allotment is similar to Int-serv in working. However, the diff-serv is developed to offer scalable as well as elastic service differentiation (Terrasa, Saez, & Vila, 2003). Part 2- H.323. 1. Describe in your own terms the four main components of an H.323 system H.323 network encompasses various components such as Gatekeepers, H.323 Terminals, Multipoint Control Units (MCUs) and Gateways. In addition, all these components work mutually through standards based video conferencing structure (21st Century Video Ltd., 2010). H.323 Terminals: The initial components of the H.323 terminals are the endpoints on the local area network that offer real-time two way interaction and communications. The H.323 standard outlines that the entire H.323 Terminals have to up hold voice, video as well as data being optional (Willamowius, 2009) & (Network World, Inc., 2010). Gatekeepers: The next component of the H.323 network is a standard that is acknowledged as the Gatekeeper, as a possible part, it is an important tool for describing as well as controlling video and voice communications throughout IP network (Willamowius, 2009) & (Network World, Inc., 2010). Interconnected Gatekeeper Zones: Third component of the H.323 network is Interconnected Gatekeeper Zone. Here Gatekeeper describes the zone as well as manages the registered endpoints inside a network. To call an endpoint inside the similar zone, we only dial that endpoints H.323 user number (Willamowius, 2009) & (Network World, Inc., 2010). Gateways: Forth component of a H.320 and H.323 network systems is Gateway, which offers conversion between circuit-switched networks ISDN and packet-based networks LAN, allowing the endpoints to correspond (Willamowius, 2009) & (Network World, Inc., 2010). 2. How do H.323 gatekeepers improve gateway functionality? A gatekeeper offers call control services to network’s registered endpoints. However, in many ways, an H.323 gatekeeper performs like a virtual switch, as well as it manages bandwidth and address translation. The set of the entire gateways, terminals as well as MCUs managed through a single gatekeeper is acknowledged as an H.323 Zone. Moreover, a gatekeeper is a logical entity that could be physically incorporated in gateways as well as MCUs (Top Bits, 2010) (cisco, 2010). 3. Describe 3 different problems or shortcomings associated with H.323. Here are the three different problems regarding the H.323: H.323 based network is limited to multimedia conferencing; consequently the complexity of the system is forced accordingly (Taylor & Hettick, 2002). H.323 is recognized for quite high connection set-up latencies, complex signaling as well as for implementation complexities (Pracht, 2009). Another main problem in H.323 protocol is its lack of scalability. H.323 is typically employed in small LANs (ingate, 2009) In case struggling to position a user over numerous domains, loops can occur. H.323 has no facility for loop detection that can create a huge network overload (Taylor & Hettick, 2002). Part 3- SIP Entities 1. Describe 5 logical SIP entities. Provide at least 3 sentences for each entity description. A SIP network is made of 5 kinds of logical SIP entities and each entity has particular participants and functions in SIP communication like a client (starts requests), like a server (reacts to intended requests), or as both. In addition, one “physical device” can incorporate the functionality of more than one logical SIP entity. For instance, a network server is able to work like a proxy server and a registrar at the same time. Here are following kinds of logical entities of SIP (Radvision.com, 2009): USER AGENT: In SIP, a UA (or User Agent) is the network’s endpoint entity. User Agents start as well as end sessions through exchanging requests as well as responses. RFC 2543 describes the user agent like an application that holds together a user and user agent client (Radvision.com, 2009). Agent server: Agent server is a server application that links the user when a SIP demand is received and also sends back a reply on behalf of the user (Radvision.com, 2009). PROXY SERVER: A Proxy server is a mediator entity that performs like both a client as well as a server for fulfilling requests on behalf of further clients. In addition, the requests are examined either inside or through passing them on, probably following translation, to additional servers. In this scenario, a proxy understands and, if essential, rewrites a request message before it is sent to the clients (Radvision.com, 2009). REDIRECT SERVER: In logical SIP entities a redirect server is a server that recognizes a SIP demand, maps the SIP address. It is different from proxy servers, since they do not bypass the requests to the additional servers (Radvision.com, 2009). REGISTRAR: In logical SIP entities a registrar is a server that recognizes REGISTER requests for the reason of renewing a location database using the contact information of the user stated in the request (Radvision.com, 2009). Part 4- SIP Authentication 1. Explain three types of SIP authentication. Explain some of the basic processes each authentication type provides, and also list any problems or weaknesses associated with each authentication type. Here we have three types of SIP authentication (Vesterinen, 2006): HTTP digest authentication in SIP: In SIP terms, HTTP digest method is acknowledged as SIP authentication. Formerly, HTTP digest is a challenge-response protocol, in that a nonce value is employed in challenging the objective. However, the response encompasses a checksum of the password, username, HTTP method, nonce value as well as requested URI. Using S/MIME in SIP for authentication; the safe S/MIME or Multipart Internet Mail Extensions in SIP is employed to carry replicates of SIP header fields within a MIME body. In addition, this facilitates authentication by signing the simulated header fields to confirm the individuality of the sender. Authentication scheme for a trusted SIP domain; trusted SIP domain authentication is another type of authentication in SIP. However, its purpose is to authenticate client user through the proxy server. The client that starts a call communicates immediately with that outbound proxy server that involves the authentication of the user client that has to be managed through that outbound proxy. Part 5- CODECs. 1. Provide a strong definition of the term CODEC, and explain how are CODECS utilized with VOIP. A codec (or Coder/Decoder) changes analog signals into a digital bit-stream. In the VoIP environment, codecs are employed to encode voice for transmission all the way through the IP networks. In addition, the codecs that are utilized by VoIP are acknowledged as vocoders, since they are designed for "voice encoding". Moreover, Codecs offer a compression facility to save bandwidth of a network. A number of codecs as well up-hold silence suppression, wherever silence is not transmitted (ozvoip, 2010) & (Codec Summary table, 2009) & (Unuth, 2009). Figure 1- VoIP CODEC Source; [http://www.topbits.com/voip-codecs.html] 2- Which two CODECs are most commonly used for VOIP Internet transmissions? Two most commonly codecs are G.711 and G.729. The codec G.711 divides the audio data into the samples at 64,000 times a second. In addition, it changes every minute data sample into digitized data as well as compacts it for transmission. The next most populer codec is 729A that has the capability of sampling audio data at the rate of 8,000 times per second as well as is the most frequently used codec in VoIP (Valdes & Roos, 2009) & (VoIP Solution Provider, 2009). Bibliography 21st Century Video Ltd. (2010). Terminals, Gatekeepers, Gateways & MCUs. Retrieved April 16, 2010, from 21st Century Video Ltd.: http://www.c21video.com/h323.html cisco. (2010). Understanding H.323 Gatekeepers. Retrieved April 16, 2010, from http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800c5e0d.shtml Codec Summary table. (2009). Retrieved 04 12, 2010, from http://www.en.voipforo.com/codec/codecs.php Edwards, J. (2007, January 22). VoIP QoS: What You Need to Know. Retrieved April 16, 2010, from VoIP News: http://www.voip-news.com/feature/voip-qos-service-quality-012207/ ingate. (2009). The SIP protocol. Retrieved 04 09, 2010, from http://www.ingate.com/files/422/fwmanual-en/xa9835.html Network World, Inc. (2010). H.323. Retrieved April 16, 2010, from Network World, Inc.: http://www.networkworld.com/details/499.html ozvoip. (2010). Introduction to Codecs. Retrieved 04 12, 2010, from http://www.ozvoip.com/voip-codecs/ Pracht, S. (2009). Agilent Technologies Troubleshooting H.323 Signaling. Retrieved 04 09, 2010, from http://docs.google.com/viewer?a=v&q=cache:O8QaXhju8-8J:literature.agilent.com/litweb/pdf/5968-3642E.pdf+H.323+shortcoming&hl=en&gl=pk&pid=bl&srcid=ADGEESjQN2PeXj8U3UoZNwcH74rHcNaEN1f8nUc10xg8zfHuQ1rLFI9kDfbbCDzBvc5iX4Mi4SNHYEwWrFlRqCepf6RAoMqF-Lggi_cSHvS0 Radvision.com. (2009). SIP: Protocol Overview. Retrieved 04 09, 2010, from http://docs.google.com/viewer?a=v&q=cache:LWx53S0lW50J:www.radvision.com/NR/rdonlyres/51855E82-BD7C-4D9D-AA8A-E822E3F4A81F/0/RADVISIONSIPProtocolOverview.pdf+SIP+logical+entities&hl=en&gl=pk&pid=bl&srcid=ADGEESgOEHlhUAGCQsFS2n_Zscx28878I0LnLApV5XAUOPu6T7Z Taylor, S., & Hettick, L. (2002). H.323 vs. SIP. Retrieved April 16, 2010, from http://www.networkworld.com/newsletters/converg/2002/01416213.html Terrasa, S., Saez, S., & Vila, J. (2003). Comparing the utilization bounds of IntServ and DiffServ. Retrieved April 15, 2010, from Department of Computing, University of Bradford: http://www.comp.brad.ac.uk/het-net/HET-NETs04/CameraPapers/P10.pdf Top Bits. (2010). Understanding the ISA Server H.323 Gatekeeper. Retrieved April 16, 2010, from http://www.topbits.com/configuring-the-isa-server-h323-gatekeeper.html Unuth, N. (2009). What is a Codec? Retrieved 04 12, 2010, from http://voip.about.com/od/voipbasics/a/codecdef.htm Valdes, R., & Roos, D. (2009). How VoIP Works . Retrieved 04 12, 2010, from http://communication.howstuffworks.com/ip-telephony6.htm Vesterinen, P. (2006). User authentication in SIP. Retrieved April 16, 2010, from http://docs.google.com/viewer?a=v&q=cache:dAQ1IBBF-eQJ:www.tml.tkk.fi/Publications/C/22/papers/Vesterinen_final.pdf+SIP+authentication&hl=en&gl=pk&pid=bl&srcid=ADGEESjfkZuG5jnUgxk-U5rOV_j-aoAZvSpayUXgox80_KXiUbflqMsKCyGZMo53RByJjhbtAjjZU21lRRhxOHY_8vKjuys VoIP Solution Provider. (2009). VoIP Audio Codecs VoIP Quality. Retrieved April 16, 2010, from http://www.voip-solution-provider.net/VOIP-quality-audio-codecs.php Willamowius, J. (2009, December 21). OpenH323 Gatekeeper - The GNU Gatekeeper. Retrieved April 15, 2010, from GNUU Gatekeeper.org: http://www.gnugk.org/ Read More