24-05-2014, 10:53 AM
Voice over Internet Protocol (VoIP)
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ABSTRACT
This paper is the first in a three part series that will ultimately detail the past, present and future of Voice over Internet Protocol (VoIP). The purpose of this paper is to detail the history of VoIP and explore this technology/industry by examining its technological history, cultural history and its economic history. For the sake of brevity, this paper (and the two that follow) will focus on this technology and its place in the business world. The goals for the second paper will be to focus on the present condition of VoIP in the business world and to draw connections to historic events. In a similar structure, the
second paper will also focus on the technology, cultural and economic factors that define VoIP in current terms. In addition, the second paper will detail particular legal or ethical issues faced by the industry, the target audience for this industry, and will present the factors leading to an explanation the digital divide.
The goals for the third paper will focus on the future possibilities of VoIP in the business world and will draw connections from both the past and present states of the technology/industry. That document will draw together the hypotheses presented by leaders in the industry and will also include my own analysis of the future of the technology. Furthermore, I will include hypothetical events that may affect the technology,
our culture and the industry’s economics. The use of VoIP by individual consumers was the beginning of a massive move from traditional telephone systems to a form of new media where voice and other forms of digital media could converge with an already established data network. Major advancements in the technology are the result of business development and adoption. This paper focuses on the history of VoIP and how this technology fits into the business setting.
History of voip
Now that the groundwork has been documented, we can examine the short brief
of VoIP. From most accounts, VoIP started in February of 1995 by a small company in Israel
called Vocaltec, Inc. Their product, InternetPhone, allowed one user to call another user via
their computers, a microphone and a set of speakers. Additionally, this application/product
only worked if both the caller and the receiver had the same software setup. By 1998
some entrepreneurs started to market PC-to-phone
Impact of VoIP on Networks
Although VoIP is based on packet switching technology and should be a more efficient
transport format, in actual fact, it is more inefficient. This is so due to the pervasive use
of SIP and Skype protocols, which are bandwidth hungry. VoIP adds significant traffic
load and latency to the network, especially if the network was not planned with that application in mind.
Network management is also more time consuming. For example, it becomes necessary
to differentiate the different user groups. Increased quality of service monitoring is also
needed to ensure service level agreements to subscribed customers.
Traditional PSTN Call vs. VoIP
Traditional telephony uses circuit switching technology while VoIP uses packet switching. In circuit-switched networks, network resources are dedicated to the circuit during the entire message, and the entire message follows the same path. In packetswitched networks, the message is broken into packets, each of which can take a different route to the destination, where the packets are recompiled into the original message. As such, packet switching is supposed to be a much more efficient and cost effective way of sending voice messages.
VoIP calls have different routing arrangements, such as peer to peer and those set up and
maintained by proxy servers. Proxy servers are logical intermediary entities that control 3 and process call requests on behalf of its group of user clients e.g. users on a LAN or within a company. A peer to peer VoIP call is a direct connection between two users. Party A calls party B, party B accepts the call and a VoIP session is established between the two users. Calls via proxy severs are established in two ways. In one scenario, Party A sends a call request for B to the proxy server, the proxy server sends A’s information to
B and B’s information to A, and maintains the connection for the duration of the call.
Gateways
Gateways are devices that enable communication between H.323 networks and other networks, such as PSTN or ISDN networks. If one party in a conversation is utilizing a terminal that is not an H.323 terminal, then the call must pass through a gateway in order to enable both parties to communicate.
Gateways are widely used today in order to enable the legacy PSTN phones to interconnect with the large, international H.323 networks that are presently deployed by services providers. Gateways are also used within the enterprise in order to enable enterprise IP phones to communicate through the service provider to users on the PSTN.
Gateways are also used in order to enable videoconferencing devices based on H.320 and H.324 to communicate with H.323 systems. Most of the third generation (3G) mobile networks deployed today utilize the H.324 protocol and are able to communicate with H.323-based terminals in corporate networks through such gateway devices.
Gatekeepers
A Gatekeeper is an optional component in the H.323 network that provides a number of services to terminals, gateways, and MCU devices. Those services include endpoint registration, address resolution, admission control, user authentication, and so forth. Of the various functions performed by the gatekeeper, address resolution is the most important as it enables two endpoints to contact each other without either endpoint having to know the IP address of the other endpoint.
Real-time Transport
In real-time interactive audio/video, people communicate with one another in real time.
The Internet phone or voice over IP is an example of this type of application. Video
conferencing is another example that allows people to communicate visually and orally.
Characteristics Before addressing the protocols used in this class of applications, we discuss some characteristics of real-time audio/video communication. SIP Protocol
The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. It was developed by a small team of IP protocol and cryptographic experts from Cisco and Ericsson including David Oran, David McGrew, Mark Baugher, Mats Naslund, Elisabetta Carrara, James Black, Karl Norman, and Rolf Blom. It was first published by the IETF in March 2004 as RFC 3711.
Since RTP is closely related to RTCP (Real Time Control Protocol) which can be used to control the RTP session, SRTP also has a sister protocol, called Secure RTCP (or SRTCP); SRTCP provides the same security-related features to RTCP, as the ones provided by SRTP to RTP.