The IP Multimedia Subsystem (IMS) defined by the 3GPP has been mainly developed and deployed by telephony vendors on vendor-specific hardware. Recent advances in Network Function Virtualisation (NFV) technology paved the way to virtualized hardware and telephony function elasticity. As such Telecom vendors start to embrace the cloud as a deployment platform usually selecting a privileged virtua...

Voice over IP (VoIP) introduces new ways of communication, while utilizing existing data networks to provide inexpensive voice communications worldwide as a promising alternative to the traditional PSTN telephony. SPam over Internet Telephony (SPIT) is one potential source of future annoyance in VoIP. A common way to launch a SPIT attack is the use of an automated procedure (bot), which generat...

The ubiquity and convergence of mobile telephony and the Internet leads many to believe that future isochronous telephony data will be transported in the form of Voice-over-IP (VoIP) datagrams. This paper investigates the modelling and aggregation of multiple VoIP streams. The relationship between the number of VoIP sources, output link rate and certain teletraffic metrics is investigated. Ther...

Status of This Memo This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. Abstract This document describes the applicability of the several protocols developed under the signalling transport framework. A description of the main issues regarding the use of the Stream Control Transmission Protocol (...

This paper speculates on the integration of application-level multicast with a Skype clone, an application that implements Voice over Internet Protocol (VoIP). Skype is a globally successful application of IP telephony using a peer-to-peer network. Researchers have shown how efficiently application-level multicast can be adopted by a peer-to-peer network, I now wish to speculate on how well a S...

Emergence of Internet Telephony has presented new challenges and opportunities in Speech Codec design and Development. Key challenges are quality of speech, usage of available low cost DSP/ARM and inbuilt error resilience. For instance, bandwidth of Internet connection is a non stationary parameter and thus the same is expected to vary over time. Devices connected to such a network need to acco...

This paper describes a concept based on application adaptivity and Differentiated Services (DiffServ) in order to provide the Quality-of-Service (QoS) required by realtime applications. Assuming the availability of different service classes, we propose that a real-time application always selects the lowest / cheapest service class that still can meet the application requirements. The selection ...

The Internet has truly become the ubiquitous communication infrastructure, bringing many benefits to the users and making the communication over IP the predominant protocol architecture for data and multimedia transmission (Web). The regular telephone service shall, in a near future, converge to the Internet, giving place to a single integrated voice and data network. IP telephony is growing, f...

In this paper a new way of exchanging information for Voice over Internet Protocol (VoIP) service is presented. With use of digital watermarking and steganography techniques we achieve a covert channel which can be used for different purposes e.g. to improve IP Telephony security or to alternate existing protocols like RTCP (Real-Time Control Protocol). Main advantage of this solution is that i...

For Internet Telephony to be a viable alternative to the Public Switch Telephone Network (PSTN), efficient and high quality communications are required. This paper proposes an optimization algorithm that selects parameters like coding scheme, packet loss bound, and maximum link utilization level in a Voice over IP (VoIP) network. The goal is to deliver guaranteed Quality of Service (for voice) ...