Αντικείμενο της εργασίας μας, είναι η δημιουργία ενός γραφικού περιβάλλοντος για τη διαχείριση του συστήματος VOIP υπηρεσιών. Αυτό είναι ο Opensips Web Manager, μια ολοκληρωμένη λύση, από άποψη διαμόρφωσης, χρήσιμο για τον διαχειριστή ενός τηλεφωνικού συστήματος. Ο OWM είναι ένα web user interface, που έχει σχεδιαστεί με PHP. Χρησιμοποιείται για τη διαχείριση των πληροφοριών του συστήματος όσο και για τη διαχείριση της κατάστασης λειτουργίας των εξυπηρετητών, που στηρίζουν ένα VOIP σύστημα. O OWM. Ο διαχειριστής έχει πρόσβαση στον OWM μέσω ενός web browser και έτσι η φυσική θέση του διαχειριστή δεν περιορίζεται από την φυσική θέση του SIP εξυπηρετητή. Η αναζήτηση πληροφοριών, που αφορούν συνδρομητές, στοιχεία κλήσεων κ.α. γίνεται με τη δημιουργία «query», προς τη βάση δεδομένων του SIP εξυπηρετητή. Η μέθοδος ανάπτυξης της εφαρμογής μας ήταν δισδιάστατη. Αρχικά έγινε η υλοποίηση συστήματος VOIP υπηρεσιών, που περιελάμβανε τις διαμορφώσεις των εξυπηρετητών DHCPs , DNSs και OpenSIPs. Ακολούθησε η εγκατάσταση του WEBs εξυπηρετητή, που φιλοξενεί τον OWM. Τέλος σχεδιάστηκε το web user interface. / The subject of our project is to create a graphical user interface for managing systems that provide VOIP services. This is the Opensips Web Manager, a integrated solution regarding configuration, useful for the administration of a telephone system. OWM is a web user interface, designed using PHP. Especially OWM is used to manage system’s data and to control server’s status, which support a VOIP system. The administrator has access to the OWM through a web browser, so the physical location of the manager is not limited by SIP server’s physical location. The search for subscriber’ s and calling feature’s data, is performed by creating «query», to SIP server’s database. The method we use to develop our application has been two-dimensional. First was the implementation of VOIP service, which included the configurations of servers like DHCPs, DNSs and OpenSIPs. Next step was the installation of a WEB server that hosts OWM. Finally, we designed the web user interface.

Magister Scientiae - MSc / SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints. / South Africa

This thesis evaluates and shows an implementation of a video telephony solution for network connected set-top boxes based on the SIP protocol for managing sessions. Unlike other video telephony implementations the set-top box does not handle both audio and video, but only video. A separate phone is used to handle audio. To maintain compatibility with other video telephony implementations, which expect a single SIP device with both audio and video capabilities, a mechanism to merge the audio (SIP-phone) and video (set-top box) into a single entity was developed using a back-to-back user agent. Due to the set-top boxes'limited hardware it could be impossible to have video compression and decompression performed by the set-top boxes. However, numerous performance tests of compression algorithms showed that the computational power available in the set-top boxes is sufficient to have acceptable frame rate and image quality in a video telephony session. A faster CPU or dedicated hardware for video compression and decompression would however be required in order to compete with dedicated video telephony systems available today. The implemented video telephony system is based on open standards such as SIP, RTP and H.261, which means interoperability with other video telephony implementations, such as Microsoft's Windows Messenger 4.7, is good.

In the thesis the advantages with Text over IP (ToIP) is explained and it is motivated why it is a good idea to integrate this in Asterisk. It also presents an implementation of a ToIP extension in Asterisk. ToIP means communicating over a network based on Internet protocols with real-time text. Real-time text means a character is sent to the receiving terminal as soon the sender has typed it or with a small delay. In the thesis IM and ToIP is compared in a survey. The result point at IM is not better than ToIP even though it is much more commonly used. VoIP can not replace ToIP either because there are occasions when ToIP is better for instance if the person using it is deaf or if a person want to make a private conversation in a noisy room. Asterisk is an IP-PBX. PBX stands for Private Branch Exchange which means a private telephone system which is part of a larger network system that exchange information. An IP-PBX is a PBX based on the Internet. Asterisk and many other IP-PBX can also exchange calls between the PSTN ant the Internet. By including ToIP in Asterisk it will be possible to exchange ToIP calls. The implementation described is not only including ToIP in Asterisk but also a translation function between the text format called t140 and another text format called t140 with redundancy. The idea is to extend the translation function in the future to more text formats.

Voice over IP (VoIP) services based on Session Initiation Protocol (SIP) has gained much attention as compared to other protocols like H.323 or MGCP over the last decade. SIP is the most favorite signaling protocol for the current and future IP telephony services, and it‘s also becoming the real competitor for traditional telephony services. However, the open architecture of SIP results the provided services vulnerable to different types of security threats which are similar in nature to those currently existing on the Internet. For this reason, there is an obvious need to provide some kind of security mechanisms to SIP based VOIP implementations. In this research, we will discuss the security threats to SIP and will highlight the related open issues. Although there are many threats to SIP security but we will focus mainly on the session hijacking and DoS attacks. We will demonstrate these types of attacks by introducing a model/practical test environment. We will also analyze the effect and performance of some the proposed solutions that is the use of Network Address Translation (NAT), IPSec, Virtual Private Networks (VPNs) and Firewalls (IDS/IPS) with the help of a test scenario.

Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more. The 3GPP (3rd Generation Partnership Project) suggests SIP as the signalling solution for 3rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols. The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1]. This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1]. The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.

With the standardization of IP Multimedia Subsystem (IMS), the telecommunications scene becomes more and more converged and in the future we will most likely access our services from all kinds of devices and link them together. One important future access method that has so far been left out of the standardization is television. There is a need for Internet Protocol Television (IPTV) to work together with IMS and this thesis focuses on one aspect of that convergence, namely charging. The problem explored in this thesis is if there is an efficient way of charging for IPTV services while taking advantage of the IMS charging functionality and this is done for two aspects of the problem. First, the possiblilty of an efficient Session Initiation Protocol (SIP) signaling schema is investigated and then a good charging Application Programming Interface (API) to be used when developing applications is investigated. The findings of these two investigations are then tested and improved during the implementation of a demo application. This thesis delivers specifications for a signaling schema that enables a Set-Top Box (STB) to pass charging information to an IMS network via INFO requests inside a special charging session. The schema is small and extendable to ensure that it can be modified further on if necessary. The thesis also delivers an encapsulating and intuitive charging API to be used by developers who want to charge for their services.

This report presents an investigation of the possibilities to implement voice over IP (VoIP) in Sony Ericsson cellular phones. The results from this investigation show that it is partially possible to implement such a solution. The best option for doing so is to make use of the support for the Session Initiation Protocol and the Real-time Transport Protocol offered by the architecture. Another goal is to evaluate if Bluetooth is able to handle the requirements needed for the solution. The whole concept is proven by implementing a prototype. Measurements on this prototype show that Bluetooth will be able to handle the requirements of most IP-based voice communication, i.e., in respect to latency and bandwidth.

Traditional public switched telephone networks (PSTN) are replaced more and more by VoIP services these days.  Although it is good for saving costs, the disadvantage of this development is that VoIP networks are less secure than the traditional  way of transmitting voice. Because VoIP networks are being deployed in open environments and rely on other network  services, the VoIP service itself becomes vulnerable to potential attacks against its infrastructure or other services  it relies on. This thesis will present a discussion of security issues of the Session Initiation Protocol (SIP), the signalling protocol for  VoIP services. The main focus is on active attacks against the protocol that aim to reduce the service's availability -- so called  Denial of Service (DoS) attacks. Existing countermeasures and detection schemes do not adequately differentiate between DoS attacks. However, the differentiation  is important with respect to performance loss, as various protection schemes involve more computationally intensive processes. Based on that discussion, this thesis attempts to provide an ontological approach to describing, and eventually preventing attacks from  having their intended effects.

Telephony plays a fundamental role in our society. It enables remote parties to interact and express themselves over great distances. The telephone as a means of communicating has become part of every day life. Organisations and industry are now looking at Voice over IP (VoIP) technologies. They want to take advantage of new and previously unavailable voice services. Various interested parties are seeking to leverage the emerging VoIP technology for more flexible and efficient communication between staff, clients and partners. <o>VoIP is a recent innovation enabled by Next Generation Network (NGN). It provides and enables means of communication over a digital network, specifically the Internet. VoIP is gaining wide spread adoption and will ultimately replace traditional telephony. The result of this trend is a ubiquitous, global and digital communication infrastructure. VoIP, however, still faces many challenges. It is not yet as reliable and dependable as the current Public Switched Telephone Network (PSTN). The employed communication protocols are immature with many security flaws and weaknesses. Session Initiation Protocol (SIP), a popular VoIP protocol does not sufficiently protect a users privacy. A user’s information is neither encrypted nor secured when calling a remote party. There is a lack of control over the information included in the SIP messages. Our specific concern is that private and sensitive information is exchanged over the public internet. This dissertation concerns itself with the communication path chosen by SIP when establishing a session with a remote party. In SIP, VoIP calls are established over unknown and untrusted intermediaries to reach the desired party. We analyse the SIP headers to determine the information leakage at each chosen intermediary. Our concerns for possible breach of privacy when using SIP were confirmed by the findings. A user’s privacy can be compromised through the extraction of explicit private details reflected in SIP headers. It is further possible to profile the user and determine communication habits from implicit time, location and device information. Our research proposes enhancements to SIP. Each intermediary must digitally sign over the SIP headers ensuring the communication path was not be altered. These signatures are added sequentially creating a chain of certified intermediaries. Our enhancements to SIP do not seek to encrypt the headers, but to use these intermediary signatures to reduce the risk of information leakage. We created a model of our proposed enhancements for attaching signatures at each intermediary. The model also provides a means of identifying unknown or malicious intermediaries prior to establishing a SIP session. Finally, the model was specified in Z notation. The Z specification language was well suited to accurately and precisely represent our model. This formal notation was adopted to specify the types, states and model behaviour. The specification was validated using the Z type-checker ZTC. Copyright / Dissertation (MSc)--University of Pretoria, 2010. / Computer Science / unrestricted