630 research outputs found
Notes on the use of RTP for shared workspace applications
The Real-time Transport Protocol, RTP, has become the dominant protocol for streaming audio and video in IP-based environments. A number of proposals have been made which attempt to build on this success and apply RTP for shared workspace applications. We discuss the needs of such applications and the features provided by RTP, with an aim to showing why RTP is not appropriate for such uses
New security and control protocol for VoIP based on steganography and digital watermarking
In this paper new security and control protocol for Voice over Internet
Protocol (VoIP) service is presented. It is the alternative for the IETF's
(Internet Engineering Task Force) RTCP (Real-Time Control Protocol) for
real-time application's traffic. Additionally this solution offers
authentication and integrity, it is capable of exchanging and verifying QoS and
security parameters. It is based on digital watermarking and steganography that
is why it does not consume additional bandwidth and the data transmitted is
inseparably bound to the voice content.Comment: 8 pages, 4 figures, 1 tabl
RTP MIDI : Recovery Journal Evaluation and Alternative Proposal
An RTP payload for MIDI commands is under development. As a part of this draft, a default resiliency mechanism for the transport over lossy networks defines a journalling method called recovery journal. But the theoretical size of this recovery journal can be very large and its format is complex. This report will present an empirical evaluation of the recovery journal size based on a few MidiFiles. We will also propose an alternative solution for the resiliency of RTP MIDI streams based on the combined use of redundancy and retransmissions. Our solution is simpler and might be interesting for some scenarios, typically: short grouping times, complex streams or unconventional semantics
A comparative study of RTC applications
Real-Time Communication (RTC) applications have become ubiquitous and are nowadays fundamental for people to communicate with friends and relatives, as well as for enterprises to allow remote working and save travel costs. Countless competing platforms differ in the ease of use, features they implement, supported user equipment and targeted audience (consumer of business). However, there is no standard protocol or interoperability mechanism. This picture complicates the traffic management, making it hard to isolate RTC traffic for prioritization or obstruction. Moreover, undocumented operation could result in the traffic being blocked at firewalls or middleboxes. In this paper, we analyze 13 popular RTC applications, from widespread consumer apps, like Skype and Whatsapp, to business platforms dedicated to enterprises - Microsoft Teams and Webex Teams. We collect packet traces under different conditions and illustrate similarities and differences in their use of the network. We find that most applications employ the well-known RTP protocol, but we observe a few cases of different (and even undocumented) approaches. The majority of applications allow peer-to-peer communication during calls with only two participants. Six of them send redundant data for Forward Error Correction or encode the user video at different bitrates. In addition, we notice that many of them are easy to identify by looking at the destination servers or the domain names resolved via DNS. The packet traces we collected, along with the metadata we extract, are made available to the community
Congestion Control using FEC for Conversational Multimedia Communication
In this paper, we propose a new rate control algorithm for conversational
multimedia flows. In our approach, along with Real-time Transport Protocol
(RTP) media packets, we propose sending redundant packets to probe for
available bandwidth. These redundant packets are Forward Error Correction (FEC)
encoded RTP packets. A straightforward interpretation is that if no losses
occur, the sender can increase the sending rate to include the FEC bit rate,
and in the case of losses due to congestion the redundant packets help in
recovering the lost packets. We also show that by varying the FEC bit rate, the
sender is able to conservatively or aggressively probe for available bandwidth.
We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network
simulator and in the real-world and compare it to other congestion control
algorithms
Non-Repudiation in Internet Telephony
We present a concept to achieve non-repudiation for natural language
conversations over the Internet. The method rests on chained electronic
signatures applied to pieces of packet-based, digital, voice communication. It
establishes the integrity and authenticity of the bidirectional data stream and
its temporal sequence and thus the security context of a conversation. The
concept is close to the protocols for Voice over the Internet (VoIP), provides
a high level of inherent security, and extends naturally to multilateral
non-repudiation, e.g., for conferences. Signatures over conversations can
become true declarations of will in analogy to electronically signed, digital
documents. This enables binding verbal contracts, in principle between
unacquainted speakers, and in particular without witnesses. A reference
implementation of a secure VoIP archive is exhibited.Comment: Accepted full research paper at IFIP sec2007, Sandton, South Africa,
14-16 May 200
Online Classification of RTC Traffic
Real-time communication (RTC) platforms have become increasingly popular in the last decade, together with the spread of broadband Internet access. They are nowadays a fundamental means for connecting people and supporting the economy, which relies more and more on forms of remote working. In this context, it is particularly important to act at the network level to ensure adequate Quality of Experience (QoE) to users, where proper traffic management policies are essential to prioritize RTC traffic. This, in turn, requires in-network devices to identify RTC streams and the type of content they carry. In this paper, we propose a machine learning-based application to classify, in real-time, the media streams generated by RTC applications encapsulated in Secure Real Time Protocol (SRTP) flows. Using carefully tuned features extracted from packet characteristics, we train a model to classify streams into an ample set of classes, including media type (audio/video), video quality and redundant streams. To validate our approach, we use traffic from more than 88 hours of multi-party meeting calls made using the Cisco Webex Teams application. We reach an overall accuracy of 97% with a light-weight decision tree model, which makes decisions using only 1 second of traffic
Evaluation of cross-layer reliability mechanisms for satellite digital multimedia broadcast
This paper presents a study of some reliability mechanisms which may be put at work in the context of Satellite Digital Multimedia Broadcasting (SDMB) to mobile devices such as handheld phones. These mechanisms include error correcting codes, interleaving at the physical layer, erasure codes at
intermediate layers and error concealment on the video decoder. The evaluation is made on a realistic satellite channel and takes into account practical constraints such as the maximum zapping time and the user mobility at several speeds. The evaluation is done by simulating different scenarii with complete protocol stacks. The simulations indicate that, under the assumptions taken here, the scenario using highly compressed video protected by erasure codes at intermediate layers seems to be the best solution
on this kind of channel
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