Detta examensarbete utfördes hos företaget Syntronic i mjärdevi. Uppgiften de ville ha utförd var en undersökning och implementation av ny funktionalitet i en redan existerande android applikation som de har utvecklat. Uppgiften i denna applikation var att undersöka möjligheterna till IP-telefoni, eller VOIP som det kallas, så att de kunde ringa varandra via applikationen genom datanätet istället för det ordinära nätet, samt att implementera detta. Detta examensarbete utfördes hos företaget Syntronic i mjärdevi. Uppgiften de ville ha utförd var en undersökning och implementation av ny funktionalitet i en redan existerande android applikation som de har utvecklat. Uppgiften i denna applikation var att undersöka möjligheterna till IP-telefoni, eller VOIP som det kallas, så att de kunde ringa varandra via applikationen genom datanätet istället för det ordinära nätet, samt att implementera detta. För att möjliggöra detta har dels android API för SIP, samt en Asterisk server som tar hand om SIP samtalen och SIP-adresserna används. Utöver detta finns det en SOA-tjänst som tar hand om databashanteringen för applikationen.

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.<p>This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. <p>The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration.

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.<p>This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. <p>The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration.

This record of study documents the experience acquired during my internship at Sonus Networks, Inc. for the Doctor of Engineering Program. In this record of study, I have surveyed and analyzed the current standardization status of Voice over Internet Protocol (VoIP) security and proposed an Internet draft on secure retargeting and response identity. The draft provides a simple and comprehensive solution to the response identity, call recipient identity and intermediate server retargeting problems in the Session Initiation Protocol (SIP) call setup process. To support product line development and enable product evolution in the quickly growing VoIP market, I have proposed a generic development framework for SIP application servers. The common and open architecture of the framework supports multiple products development and facilitates integration of new service modules. The systematical reuse of proven software design and implementation enables companies to reduce the development cost and shorten the time-to-market. As the development and diffusion of VoIP can never be isolated from the social sphere, I have investigated the current status, influence and interaction of three most important factors: standardization, market forces and government regulation on the development and diffusion of VoIP. The worldwide deregulation and market privatization have caused the transition of the standards development model. This transition in turn influences the market diffusion. Other than standardization, market forces including customer needs, the revenue pressure on carriers and vendors, competitive and economic environment, social culture and regulation uncertainties create both threats and opportunities. I have examined market drivers and obstacles in the current VoIP adoption stage, analyzed current VoIP market players and their strategies, and predicted the direction of VoIP business. The regulation creates the macro environment in which VoIP develops and diffuses. I have explored modern telecommunications regulation principles based on which government makes decisions on most current issues, including 911 support, mergers and acquisitions, interconnection obligation and leasing rights, rate structure and universal service fees.

With the popularity of the VoIP, the method of communication is gradually changing from email to internet telephony. The email communication is always being used for advertising and phishing. Also, this junk information may be used for internet telephony and may disturb users while the internet telephony becomes more and more popular. SPIT (Spam for Internet Telephony)may become a serious problem in the network. Moreover, the influence cause by internet telephony maybe more serious than email. Hence, this paper wants to provide a SIP terminal that can filter the caller. With limited resources, the filter uses white and black lists and integrated with back-end SPIT database to design an efficient SPIT filter. Through this filter, we can block the spam call efficiently and keep the quality of the VoIP service.

VoIP applications become popular in recently years, and more and more network researchers are interested in VoIP issues. However, users are used to using the traditional telephony number to make calls. This is because the telephony number is more convenient than the meaningless IP address and the diffusive domain name in VoIP networks. They are really hard to remember. In this paper, we discuss the design and implementation of ENUM (tElephone NUmber Mapping) service over a SIP user agent. According to the technology of ENUM (RFC2916), the dialed numbers can be encoded into E.164 numbers. By querying the DNS, we can get the SIP URI (Universal Resource Identifier) which exists in the NAPTR (Naming Authority Pointer) resource records. The ENUM service plays an important role between the traditional telephony networks and the VoIP networks. The users can be unaware of the different operation style in VoIP.

With the rapid growth of communication, various communication equipments get rid of the stale and bring forth the fresh. The frequently used communication device is not limited to domestic telephone any longer. We can achieve the same functionality as traditional telephone by the use of VoIP¡]Voice over IP¡^over the Internet. The common fraction of these communication products is the user identity which may be achieved by either a telephone number or an URI (Universal Resource Identity). In this paper, we implement a distributed system with load balancing. This is achieved by utilization of ENUM. It can distribute the load of the traditional server to several servers to enhance the reliability of services. For the future combination of VoIP and the traditional PSTN, we expect to provide more consistent services while the number of user is increasing.

VoIP technology is one of the important applications of the network. In addition to using traditional PSTN telephone, people can choose favorable VoIP telephone because of the Internet popularization and High speed Internet. RTP (Real-time Transfer Protocol) is the foundation of VoIP, and it is suitable for applications to manage real-time transmission of multimedia data. But RTP does not provide any confidentiality, the eavesdrop of VoIP is easy to realize. In this paper, we discuss how to design and implement an embedded SIP user agent that provide an fast secure encoding of RTP voice streaming. We use limited resources to implement RTP encoding mechanism to prevent anyone from eavesdropping .

The goal of the master’s thesis was to propose two laboratory exercises, integrating newly purchased devices HTC Desire and Nokia N900. Designed tasks bring new technologies and services to education process. The first task examines the configuration of branch exchange Asterisk PBX and analyzes SIP protocol. Part of exercise is concentrated on high-speed data transmission in mobile and wireless networks, regarding to usability of VoIP technology. The second exercise introduces to vulnerability of VoIP technology. It contains simulations of attacks on branch exchange Asterisk, DoS attack and discusses methods to secure VoiP communication. The part of this exercise examines usage of HTC Desire phone, instead of ordinary Wi-Fi access point.

Voice over IP (VoIP) and wireless are revolutionary technologies by all means of modern time which change the attributes of communications dramatically. VoIP has been established as potential alternative to tradition public switched telephone network (PSTN) technology whereas Wireless communication is the most widely used access method where fixed or remote access to network resources is important. Since both technologies have shown their existence in today’s market individually, merger of these technologies was necessary and hence both technologies are being deployed but the question is whether these newly merged combination of technologies will be able to serve up to the same level of expectations and survive in current economic downturn and helps the companies to cut their cost drastically or will be finish with the time. This thesis explores the main factors affecting the deployment of VoIP over Wireless by the telecom operators and its adoption of VoIP over Wireless by Small Medium Enterprises (SMEs) and general consumers. It includes QoS, Security, user behavior, regulations, and last but not least current economic downturn. It would be a difficult task to explain all these issues with regards to the world market in this thesis so throughout our research, European market will be our preliminary focus. First, the thesis will briefly look into the future services, applications and trends which will be beneficial in supporting the growth rate of VoIP over Wireless in near future and then we will investigate all those trouble making factors causing a slow adoption rate of VoIP over Wireless. At the end of the thesis, there will be some proposals and suggestions to improve the growth rate and adoption of VoIP over Wireless