نتایج جستجو برای: voice over internet protocolvoip
تعداد نتایج: 1314036 فیلتر نتایج به سال:
Voice Over Internet (VOIP) is a rising popular Internet application that offers good services by Mobile Ad-hoc Network (MANET). MANET offers a appropriate platform for the deployment of multimedia and voice session over IP network in various application scenarios that provides safety to comfort related services. But this network also faces many issues on QoS because of packet drop, due to trans...
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...
Speech quality estimation of voice over internet protocol codec using a packet loss impairment model
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) ...
This document defines a Session Initiation Protocol (SIP) event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions. Voice call quality information derived from RTP Control Protocol Extended Reports (RTCP-XR) and call information from SIP is conveyed from a User Agent (UA) in a session, known as a reporter, to a ...
This document defines a Session Initiation Protocol (SIP) event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions. Voice call quality information derived from RTP Control Protocol Extended Reports (RTCP-XR) and call information from SIP is conveyed from a User Agent (UA) in a session, known as a reporter, to a ...
Auracle is a “group instrument,” controlled by the voice, for real-time, interactive, distributed music making over the Internet. It is implemented in the Javaprogramming language using a combination of publicly available libraries (JSyn and TransJam) and custom-built components. This paper describes how the various pieces — the voice analysis, network communication, and sound synthesis — are i...
This paper presents a non-intrusive method of determining network performance parameters for voice packet flows within a VoIP (Voice over IP, or Internet Telephony) call. An advantage of the method is that it allows not only end-to-end performance monitoring of flows, but also makes it possible to inspect the transport parameters a specific network or link when delay sensitive traffic transits ...
Auracle is a voice-controlled, networked sound instrument which enables users to control a synthesized instrument with their voice and to interact with each other in real time over the Internet. This paper describes the architecture of the system in detail, including the multi-level analysis of vocal input, the communication of that analysis data across the network, and the mapping of that data...
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