The MGCP or Media Gateway Control Protocol designed exclusively for telephony, gateway is another VoIP protocol standard that provides guidelines for control, signalling, and processing skills on telephony gateways. It is a master/slave protocol where commands sent by call agents that are implementing the signalling layers of H.323 are executed (Kaeo, 2003, p.186). Defined in RFC 3261, the SIP or Session Initiation Protocol, initial published in 2001 for the purpose of altering and terminating IP sessions in a network (Yongfeng et al. 2008, p.1) is considered the primary IETF standard for multimedia conferencing over IP.
Unlike H.323 that uses a more traditional circuit-switched approach to signalling, SIP uses ASCII codes that can be use to set up, maintain, and terminate call between two or more endpoints within the application layer control protocol. SIP is part of the IETF multimedia data and control architecture that includes RTP protocols for transporting and providing real-time data and quality of service feedback respectively (Kaeo 2006, p.187). 3. Security Issues in Implementing VoIP Vulnerabilities if exploited can severely harm a system or a network and these weaknesses are not unique to VoIP but affect all networks carrying voice transmissions, regardless of medium.
The security of an IP telephone-based network like VoIP depends on a large number of components that include the computer and its operating system, software, modem, telephone, ISPs, etc. Threats to VoIP may be inherited from conventional telephone systems or may come from those components associated with the IP networks or VoIP specific protocols (Douligeris et al. 2007, p.229). Some of the most important security requirements of a VoIP service are integrity, privacy, authenticity, availability or protection from DoS attacks (Davidson et al. 2006, p.221; Gomathi & Bhagyaveni 2008, p.1). Integrity is essential particularly in VoIP signalling since it is critical for a recipient to receive the packets without any alteration.
A third party must not be able to modify the packets in transit and similarly, they should not be able to read the data as it can compromise the sender’s and recipient’s privacy. In addition, both parties (sender & recipient) must be authenticated to ensure that the peer they are communicating is the real one. More importantly, the service must be available at all times thus DoS attacks should be prevented (Davidson 2006, p.221). In order for a network to have confidentiality, the information contained, transformed, or transported by that system cannot be read or retrieved by unauthorized entities.
The integrity attribute provides reasonable certitude that information contained, transformed, or transported by a system has not been modified by unauthorised entities while in containment, transformation, and in transport. The availability attributes provides a reasonable certitude that information contained, transformed, or transported by a system is at hand and provides a high rate of dependability (Nichols & Lekkas 2006, p.435; Bauman et al. 2006, p.12). IP telephony security risks of data network include attacks that reduce or compromise the functionality of a software system via a buffer or bandwidth overrun.
The DoS or Denial of Service occurs when an attacker create certain conditions within a network using specific codes that would trigger a denial of service. There are also types of attack where a third party can monitor, record, block, or alter data transmission and autonomous software that can travel across the Internet and IP networks and infect vulnerable host by replicating themselves. Access control on a VoIP network is often the main concern since limiting access in the name of security may also result in a poor experience or a long-term resentment by users (Wallingford, 2005, p.
223; Materna 2006, p.1)). Normally, security and privacy concerns are not often associated with communications but this is not the case with VoIP.
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