Options for Securing RTP Sessions

The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

Reflections on security options for the real-time transport protocol framework

The Real-time Transport Protocol (RTP) supports a range of video conferencing, telephony, and streaming video ap- plications, but offers few native security features. We discuss the problem of securing RTP, considering the range of applications. We outline why this makes RTP a difficult protocol to secure, and describe the approach we have recently proposed in the IETF to provide security for RTP applications. This approach treats RTP as a framework with a set of extensible security building blocks, and prescribes mandatory-to-implement security at the level of different application classes, rather than at the level of the media transport protocol

Status of This Memo

This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards " (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2006). This document specifies how to establish secure connection-oriented media transport sessions over the Transport Layer Security (TLS) protocol using the Session Description Protocol (SDP). It defines a new SDP protocol identifier, ’TCP/TLS’. It also defines the syntax and semantics for an SDP ’fingerprint ’ attribute that identifies the certificate that will be presented for the TLS session. This mechanism allows media transport over TLS connections to b

Options for Securing RTP Sessions

The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism

The Virtual Device: Expanding Wireless Communication Services Through Service Discovery and Session Mobility

We present a location-based, ubiquitous service architecture, based on the Session Initiation Protocol (SIP) and a service discovery protocol that enables users to enhance the multimedia communications services available on their mobile devices by discovering other local devices, and including them in their active sessions, creating a 'virtual device.' We have implemented our concept based on Columbia University's multimedia environment and we show its feasibility by a performance analysis

Large-Scale Measurement of Real-Time Communication on the Web

Web Real-Time Communication (WebRTC) is getting wide adoptions across the browsers (Chrome, Firefox, Opera, etc.) and platforms (PC, Android, iOS). It enables application developers to add real-time communications features (text chat, audio/video calls) to web applications using W3C standard JavaScript APIs, and the end users can enjoy real-time multimedia communication experience from the browser without the complication of installing special applications or browser plug-ins. As WebRTC based applications are getting deployed on the Internet by thousands of companies across the globe, it is very important to understand the quality of the real-time communication services provided by these applications. Important performance metrics to be considered include: whether the communication session was properly setup, what are the network delays, packet loss rate, throughput, etc. At Callstats.io, we provide a solution to address the above concerns. By integrating an JavaScript API into WebRTC applications, Callstats.io helps application providers to measure the Quality of Experience (QoE) related metrics on the end user side. This thesis illustrates how this WebRTC performance measurement system is designed and built and we show some statistics derived from the collected data to give some insight into the performance of today’s WebRTC based real-time communication services. According to our measurement, real-time communication over the Internet are generally performing well in terms of latency and loss. The throughput are good for about 30% of the communication sessions

Service composition based on SIP peer-to-peer networks

Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation