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Current Projects - Spring 2012 |
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The Voice and Video over Web Program studies systems that use the World-Wide-Web to support real-time streaming media applications. Below are descriptions of the projects in this program.
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The Voice over IIT Program is based on the VoIIT test bed. This lab test bed is an IP PBX system for the use of the IIT community and its friends and supporters. The system uses open source SIP servers and supports IP-based calls as well as calls to and from the PSTN. It also provides the following features: voice-mail; voice-mail-to-email; audio conferencing.
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The Voice and Video over Wireless Program studies systems streaming media services that are carried over wireless networks. Below are descriptions of the projects in this program.
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The ESINet Program studies the functional elements, architectures, protocols and underlying networks carrying emergency services over IP infrastructure. The goal is to characterize the performance of these networks under various conditions, and using various methods for providing security, reliability, redundancy and robustness.
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Security for real-time communications services and applications needs to be defined and analyzed. Systems that address the many axes along which real-time security can be defined need to be developed, tested and implemented in the real world. There are many facets to be considered. These include: The security of the network that is carrying the real-time application; the security of the signaling information including the identity and usage patterns of the caller and called parties; and the security of the media flows themselves. The RTC security program takes on projects that study security mechanisms and implementations, their efficacy as well as their impact on the quality of experience, and the reliability of the services and networks that they protect.
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The Session Initiation Protocol (SIP) has been adopted by many sectors of the telecommunications industry: Enterprise IP PBX es and SIP Session Border Controllers, as well as the Emergency Services IP network (ESINet) for the delivery of Emergency calls are some important examples. Many commercial systems and solutions based on SIP are available today to meet the industry s requirements. As a result, there is a strong need for a vendor-neutral benchmarking methodology to allow different SIP servers to be meaningfully compared one with the other. The goal of this program is to develop such benchmarks and to design systems to collect the benchmarks we define. The outcome of the program will include creation of two IETF drafts that describe the benchmarks and the methodology to be used for their collection. The outcome will also be a tool that collects the benchmarks and that is built in accordance with the methodology we describe. Current versions of the benchmarking methodology and terminology documents are available at http://tools.ietf.org/html/draft-ietf-bmwg-sip-bench-meth-03 and http://tools.ietf.org/html/draft-ietf-bmwg-sip-bench-term-03. The Methodology document was updated since the 03 posting. The latest version, 04, is available on the project web site under the name "draft-ietf-bmwg-sip-bench-meth-04".
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There are many commercial IP-PBX systems that offer enterprise features that are termed Unified Communications. The RTC lab has received some of these as donations and we need to configure them into stable test beds that can be used both for lab exercises and as parts of other projects. The projects in the program are focused on building and characterizing these test beds.
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Goals: This project is a study of Real-Time Media services as they are deployed today. Its scope includes services such as YouTube, HuLu, Skype, Webex, Google+ Hangouts, which are available to anyone with an Internet connection, and extends to the unified communications services and IP PBX es on offer both commercially, as for example Microsoft Lync, and through open source distributions, as for example Digium s Asterisk Server. Our goal is to analyze (1) the methods used to create these services and (2) the impact of the services on the networks - the nodes and links - that carry them, and (3) the experience of the end-user. The methods for delivery vary: Some use specialized but open client-server protocols such as SIP to set up the media streams and use other specialized protocols such as RTP to encapsulate the media in the streams; Skype uses its own proprietary peer-to-peer protocol and architecture. Many other real-time media applications use the services of HTTP to carry the media and identify the communicating parties. The resources consumed include: bandwidth, transport ports, computing cycles in the nodes, and time - measured for example in the end-to-end delays experienced by the end-user. The experience of the end-user can be characterized in a somewhat objective fashion using voice quality and video quality metrics. The individual experimenter can also subjectively indicate his/her experience during the data collection using a rating parameter.
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