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Tele-Technologies of TelerehabilitationThis section provides an overview of the telecommunication technologies that relate to telerehabilitation applications, the a primary focus on conferencing standards. ITU's Videoconferencing StandardsThe impact of teleconferencing standards by the International Telecommunications Union (ITU) has been considerable. Since introduced in roughly 1996, costs for products have come down dramatically while quality has improved. Systems have become considerably easier to use, and designed more an consumer products. These standards span both the classic phone line infrastructure (H.320 for moderate and higher bandwidth, H.324 for lower bandwidth videophones) and the packet-based Internet Protocol (IP) infrastructure (H.323, SIP). Each of these standards is really a specification of a collection of protocols, especially the H-series for audiovisual and multimedia systems (e.g. H.32x for videoconferencing, H.26x for video codecs, H.28x for remote device control, H.233-5 for security/confidentiality/encryption, H.350.x for director services architecture, H.450.x for call service features) and the G-series for audio codecs (e.g. G.72x). For a sample of conferencing solutions see the telemedical site.
Each overall videoconferencing standard defines messaging protocols that govern transmission of audio, video and data between two (or more) systems, and specifies additional protocol standards for video and voice codecs, security, privacy, and multiplexing and data control. Often they share some features, such as RTP/RTCP) for real-time messaging control and CIF (352 x 288 pixels) or QCIF (176 x 144) or the standard TV resolution of 4CIF (704 x 576). Each is periodically updated. An example of this is the recent H.264/MPEG-4 advanced video coding standard, an outgrowth of a concerted joint effort by two key international bodies developing video coding standards: the H.26x video coding standards group through ITU and the MPEG-4 computer multimedia transmission/storage standards group through ISO/IEC. It provides higher quality video at lower bit rates (same video quality at roughly half the bandwidth) and better error resilience. This has significantly improved quality-of-service, especially for broadband (e.g., LAN, cable-modem, DSL) but also for phone-based connections, and tips the scales to IP-based solutions. All of the top videoconferencing manufacturers, such as Polycom, VCON and Tandberg, are evolving to products that support both the H.323 and SIP and use the H.264 protocol. Similarly, the top digital video products used for streaming multimedia, multicasting or playing DVD movies, have or are migrating to the very flexible new MPEG-4 standard. As an example of this impact, the Mobile Usability Lab (MU-Lab) of the RERC-AMI abandoned the need to maintain protocols for “lower quality” and “higher quality” digital video storage (related to length of trials and concerns of file size), simply because the new MPEG-4 standard provides high quality storage even with relatively smaller files. Advantages of conferencing/multimedia standards:
Possible disadvantages:
It is common for companies to try to add value to their product that is over-and-above the standard. This helps explain why video and sound quality is often worse when products for different manufacturers connect during a teleconference - they connect so as to met the minimal standard, without value-added features. Our Telerehab and Performance Assessment Lab has multiple examples of each of these standards, and you have an opportunity to experience systems supporting each of these standards. See also our Glossary of Conferencing Terms. H.320 VideoconferencingThe International Telecommunication Union's (ITU's) H.320 standard is applied mostly to dedicated circuit-based switched network (point-to-point) connections of moderate or high bandwidth, such as through the medium-bandwidth ISDN digital phone protocol or a fractionated high bandwidth T1 lines.
Multipoint connections distribute equal bandwidth. For instance, with our 512Kpbs capacity, we can coordinate a five site conference in which each connection is at 128Kbps, or a three-site conference at 256Kbps (as long as each of the two other sites can handle 256Kbps). Throughout the 1990s, federal grants programs helped support the implementation of T1 hub-spoke networks to rural communities, primarily for civic, educational and health needs. These were initially very expensive, and because of lack of interoperability between major vendors each state tended to pick one company. As systems H.320 emerged, this all changed. The H.320 protocol now sees widespread use for conferencing of all types, ranging from telemedicine consultations between a tertiary and rural hospital to our routine use of H.320 for our RERC meetings. It is the pillar for most hub-spoke networks. For telehealth applications, an especially useful feature is both local and remote control of cameras -- remote control by specialists is part of the guidelines for obtaining reimbursement from the United States’ Medicare program. Virtually all of the newer H.320 products are also H.323 compliant, and can facilitate transmission of one or more channels of data (e.g., signals, images, records, presentations). This is true, for example, with our Viewstation MP systems (Polycom). H.324 VideophonesThe ITU’s H.324 protocol, intended for “videophones” that use POTS (plain old telephone service) phone lines (available in 97% of US households). Products meeting this standard are not much more difficult to use than a high-end phone or a VCR. The main user-controlled feature being interactive control of the tradeoff between refresh rate (e.g., typical range of 1 to 15 fps) and image quality. Here is a list of features:
Examples:
Our past evaluation of H.324 products (e.g., see Chapter 13 by Tran et al) suggested that while most products are interoperable (often with some effort), there is no way around the choppy nature of the transmitted video, especially if color is desirable (32,34). For home telehealth products, a small screen (e.g., 3” by 5”) is often used so that pixelation is less noticeable. Such small screens can be integrated into standard phones, with cameras integrated and/or connected by wireless means. These videophones are usually targeted towards telehomecare applications where ease-of-use is a high priority and high-quality video is not critical. There are a number of telehealth products that integrate videoconferencing with vitals, with "telenurse" and "patient" terminals that differ. There are now products that are H.324/H.323, which makes some sense given the emergence of cable modem / DSL service. IP Videoconferencing: ITU's H.323 and IETF's SIPConferencing over the packet-based circuits, often called Internet Telephony, refers to real-time transport of multimedia telephone calls over the Internet. When only voice is transferred, it is often called Voice over IP (VoIP). VoIP is currently a very big deal as we gradually transition towards an Internet-based phone system. Advantages of IP-based conferencing include low costs (e.g., web cams are as cheap as $20, and Microsoft's client-side H.323-based MSN Messenger (and NetMeeting) packages, and their SIP-based Windows Messenger product and Real Time Communication software suite within the .Net Framework, are free). Another advantage of the complex, multi-faceted standard is the support for multi-point conferences and for interfaces to H.320 systems. But perhaps the biggest advantage is the integration with computer-based data and application sharing. While an attractive alternative to circuit-based approaches such as those using H.320 or H.324, there are several fundamental disadvantages:
Currently, the most common way to implement such calls is via the H.323 protocol, and perhaps 90% of VoIP calls use this detailed ITU-T standard that comes out of the telecommunications community. A popular alternative approach, the IETF's Session Initiation Protocol (SIP), comes out of the internet software engineering community. We will discuss both. H.323 "describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 may provide real-time audio, video and/or data communication" (from ITU-T Recommendation H.323 V4). Notice the explicit mention of a lack of guaranteed quality of service. In fact, H.323 serves as an umbrella for a collection of other "best practice" standards. H.323 entities are:
An important subset of H.323 is the “voice and data” mode (i.e., video is not required for compliance, but if available must meet certain standards). To give some history, in the late 1990's, Microsoft's Netmeeting package, freely available, became a de facto "gold standard" for low-end videoconferencing over IP. It was one of 9 IP-based packages that Donal Lauderdale and myself evaluated in 1998. In addition to support for H.323, including the T.120 standard (e.g., including platform-independent support for chat, file transfer, shared white board), it provided support for application-sharing on the Windows platform. For the most part, we found the various products we evaluated to be interoperable, but typically not without some effort. In about 1998 Microsoft also embedded Netmeeting within its MSN Messenger product. While the addition of video (or applications such as powerpoint) to voice was nice, a key problem with Netmeeting was an audio time delay of about a quarter of a second. This was in part built in to their implementation of H.323, perhaps in part because of concerns of unpredictable packet delays. A summary of Microsoft's implementation of T.120 is available. The quality of video for Netmeeting and similar products was strongly a function of bandwidth, and was pretty decent for connections involving two sites on a LAN. Because Netmeeting added value to H.323 via application-sharing on Windows platforms, and also had a "minimal" implementation with an available SDK, many third-party companies provided added value to the H.323/Netmeeting protocol through features such as multi-point conferencing and more convenient phone-like calling options. Importantly, current state-of-the-art H.323-based VoIP products do not have significant audio time delays, and there are many H.323 vendors. While most H.323 implementations are proprietary, there is also an open source forum (www.openh323.org) with which our group used to participate. It is possible to access the core of the H.323 standard from the ITU site. A good source, one of many available on the web, is the H.323 Forum. The competition with the alternative SIP protocol has helped push improvements into theH.323 often-updated standard, and it is now up to Version 4 (H.323-V4), with there being many H.323 V4 products on the market and more coming. With Windows XP, Microsoft dropped continued development of H.323-based Netmeeting in favor of the SIP-based collection of standards, discussed below. Microsoft has explicitly embedded SIP within its new .Net framework, and SIP is used for its Windows Messenger product (versus H.323 for its MSN Messenger product). The former has much shorter audio time delays and more features. It is also embedded within the .Net Framework (essentially a library), and thus SIP becomes one of many tools that is available for Windows developers for platforms ranging from mobile (e.g., PocketPC) to desktop. SIP is "an application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users." (IETF RFC 2543 SIP). It consists of:
SIP (IETF FC 2543) is thus a simpler control protocol which uses textual (ASCII commands) client-server model to create, maintain, modify or terminate multimedia sessions with one or more participants. It is intended to be used as a software module for managing sessions, similar to the strategy of HTTP protocol for the web or SMTP protocol for email. To the programmer, SIP is a toolbox that turns a telephony or multimedia session into a web application that can integrate with other Internet services. In considers user location, user capabilities, user availability, call setup and call handling. A nice summary is available from the SIPCenter, and to get a sense of the degree of commercial activity see the SIPCenter main page. SIP differs from H.323 at a fundamental level: in SIP, the " intelligence" in distributed out to clients (i.e., their computers) in a more distributed architecture, as opposed to the H.323 model of an intelligent central coordinating site surrounded by "dumb" terminals. Collectively, H.323 and SIP provide a a suite of standards (see also VoIP standards Reference page at protocols.com. The current "killer application" is not videoconferencing, but VoIP, or what is now often called simply a "digital phone." Billions of dollars rest on who and what coordinates a phone connection. Many believe that there will be a gradual shift away from standard circuit-based PSTN toward an packet-based Internet system, once consumers are convinced that quality-of-service issues can be addressed. This ha largely happened. Indeed, Marquette's telephone backbone is typical: we depend on products from Cisco and Siemens, and both of these companies have products for both the H.323 and SIP standards, and well as traditional digital/analog circuit-based phones. Packet-based approaches are starting to do well at large companies with their own controlled LAN environment. Siemens in particular has come to campus to try to convince Marquette to upgrade its phone network to these new hybrid, multi-standard phones. Of note is that while most H.323 and SIP system make use of computers, there are "phone" systems that do not require the user to own a computer (e.g., mm146 from Motion Media for H.323 over cable/DSL, DV325 from 8x8 for SIP). In our own lab R&D where we've wanted to integrate our Intelligent Telerehabilitation Assistant (ITA) with videoconferencing capabilities that include mobile (PocketPC) capabilities, we originally spent some time understanding the H.323 standard, then switched to SIP. Wireless Technologies - General ConceptsReading material: part of Chapter 11 (Winters), pp. 100-103
Classification scheme being used in the IEEE wireless standards process:
Existing Medical Telemetry Products
Multi-Node Collaborative ConferencingRecent years have seen a remarkable evolution in networking capabilities, Internet use, available bandwidth and new multimedia standards. A consequence is that there is more potential for multimedia connectivity between people than was previously possible. This includes the formation of "virtual communities" for This section briefly reviews some of the approaches and terminology for such connectivity, which could potentially change the how healthcare services are delivered, and indeed the field of clinical rehabilitation. A gateway works as a mediator between IP and ISDN protocols. It addresses the occasional need for connecting calls from registered end point nodes (or terminals) to remote parties with H.320 and H.323 protocols. While perhaps less of an issue now that so many products support the H.320 and H.323 protocols, it clearly is valuable. In the Telerehab lab we have access to Gateways both through our MXM software from VCON and our TMS software from Tandberg. The process involves having the caller must first register with the gateway administrator. There are also security settings and call control considerations. To date, we've had limited experience with establishing gateways simply because the need hasn't been there. A Multipoint Control Unit (MCU) is used for connecting registered end point nodes in a multipoint videoconference. Such conferences may be ad hoc (others invited) or dedicated (pre specified, specific end points). A conferencing bridge is a structured MCU that enables initiation and management of multipoint videoconferences and often of multicast streaming conferences. It may include value-added features such as voice-activated switching (where participants see the video of the speaker whose audio signal is the strongest) or continuous presence (where certain participants are always seen and/or heard). A gatekeeper is used to connecting end points over WANs, e.g. it manages end point information like node directories and permission levels, performance tasks such as bandwidth allotment and service availability, and perhaps practical administrative functions such as event logs and billing. At the simplest level, a gatekeeper might only support IP-based connectivity. More generally, for a sophisticated gatekeeper many of the end point nodes may have different properties (e.g., mix of H.320, H.323, SIP; each with different bandwidth capacity). Often gatekeepers interact with each other. |
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