SGN1 WP 1311
04/16/2007DRAFT ACP/WGN 11/SG N1
ICAO/AERONAUTICAL COMMUNICATIONS PANEL (ACP)
Sub-working group N1
A Survey of Voice over Internet Protocol (VoIP) ApplicationsVoice over IP (VoIP) Analysis
and its Ramifications for
Air Traffic Management (ATM) Deployment
Prepared by: WGN/ Eric WeillSGN1
Presented by: Kelly Kitchens
This working paper examines current VoIP implementation efforts in government and industrythe markets, and what may be inferred from these cases regarding the feasibility of VoIP for Ground-Ground (G-G) ATM communications.
The terrestrial voice communications industry continues to has been undergoing a technologyological revolution and over the past 20last two years there, has been awith the steady transition of telephony communications lines from Time Division Multiplex (TDM) PBX to IP-based PBX and ultimately to Voice over Internet Protocols.
The motivation for this technical revolution was expected cost savings in personnel and infrastructure. These savings focused on reduction in staff requirements, network convergence and simplification, which is the combining of voice and data service over a single network infrastructure, a decrease in system costs (no dedicated proprietary telephone equipment), and easier reconfigurations (SMTP based network management). This paper will analyze discuss various aspects involved inVoIP technology for suitability for use in air traffic communications manifesting this change.
The start of telephone service and data network services were independent developments, which have merged because of the digitization of the analog voice signal. Data communications was initially a service that consisted of a low speed connection via a telephone line via Internet Service Provider (ISP).
Telephone services are based on an array of technologies. They range from analog circuit switched to digital circuit switched and fiber optic technologies, which is referred to as the public switched telephone network (PSTN). These technologies are maximized for voice services and are expensive to purchase, complicated to implement, and proprietary. Therefore costly for value added services like video and broadband data. In addition, the suite of protocols necessary to implement the various services grew cumbersome and sometimes created conflicts within the telephone network.
Large-scale deployment of digital packet data service was initially implemented using X.25 protocols. However, the low bandwidth capability and the design foundation for use of X.25 over 64k bps DS0 telecommunication links made this technology inefficient for high-speed data. This resulted in the migration towards TCP/IP protocols and services.
The developments in digital voice, the need for higher bandwidth data services coupled with the desire to simplify network topology and decrease the costs of communications led to the experimentation of digital voice signals over data networks. These experiments eventually moved into the commercial world allowing companies and network users to use the Internet as a communications infrastructure for voice communications. However, during the infancy of VoIP deployment network availability and QoS were issues. The resolution of for these problems of higher levels of reliability, availability, and quality of service were to deploy private internets to ensure levels of service similar to toll quality voice calls on a conventional telecommunications network.
The initial use of packetized voice was as a best effort service over the Internet. However, once vendors and companies relized the benefit of using data networks to avoid long distance toll charges and tarriffs an industry emerged that developed additonal data networking protocols and equipment to enable VoIP system interoperation with the PSTN system. Additionally, a new type of communications service provider was emerging that built the network infrastructure based on VoIP technology instead of circuit switched technology.As is usually the case in the commercial sector, expected cost savings generally drives infrastructure changes. In the voice communications industry, this has resulted in the migration to IP-based technologies, with VoIP now outselling digital PBX lines.
Primary factors realizing such savings from the migration towards VoIP include:
Judicious selection of product offerings, including trade-off studies between adopting a total single vendor turnkey framework, best-of-breed selection of components among vendor offerings, or open-source elements within the system
Robust planning in advance of equipment deployment and installation
Investment in management tools to monitor and control network operational parameters
Sufficient staffing undergoing a disciplined training approach to ensure proper and efficient network operations. Some of this burden may be alleviated with outsourcing, especially for small-to-medium size networks that do not require full time staffing.
The result of such measures is a reduction in staff profiling, since VoIP deployments and reconfigurations are easier to implement in the field by automation than TDM. Telecom circuit and cabling costs are reduced due to more efficient utilization of media.
Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, and Voice over Broadband is the routing of voice conversations over the public Internet or through a private IP-based network (private internet). The Protocols used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. A detailed analysis of primary VoIP technologies are contained in The Voice over IP Handbook for Air Traffic Management Applications.
Voice over IP is no longer a hobbiest or big business service. Most telephone, cable TV, and satellite TV service providers are migrating and offering to their subscribers voice over IP telephony technology. This is due to network convergence, proven technology, and cost savings. The following analysis will focus on three different aspects of voice over IP services, which are business, technical, and operational.
In the past few years the VoIP landscape has changed. The telecommnications industry is moving away from circuit-based voice solutions. The Radicati Group a telecommunications market research firm predicts that 74 percent of all corporate telphony lines will be IP-based by 20091. In addition, end of life and end of support notices for time-division multiplexing (TDM) private branch exchanges (PBXs) and spare parts availability are prompting companies to re-evaluate their voice systems.
The rapid acceptance of VoIP standards and applications are making system installations much easier than traditional voice systems. Infornectics, an international market research firm projects that worldwide revenue from IP-capable PBX equipment will reach 10.2B in 2008, up from 3.6B in 20032 .
A Nemertes Research Survey3 of 90 IT executives reports that over time there will be significant cost savings. In order to determine cost benefits of deploying VoIP hard costs like hardware, software, head count, and expenses need to be considered. Another identified cost are softcosts, which consists of commonly performed tasks like moves, adds, or changes.
Most companies researched generally catigorized savings in to the following areas:
The amont of savings or avoidance will be determined by the size of the operation, required calling area, and where the company is in their capital equipment depreciation lifecycle. A company that recently purchased telecommunications equipment may have a harder time jutifiying changing TDM equipment for VoIP equipment. Yet, the same company may be able to achieve significant savings in staffing and cost avoidance to justify migrating TDM voice traffic to VoIP services. The eventual driver will be that the TDM equipment will no longer be available. Therefore, it may be in the best interest of the company to establish a policy to limit TDM equipment purchase to”as required to sustain operation” and make all future purchases VoIP.
Another business consideration are emerging services, features that were once supplemental can be provided at no additional cost and in some cases are not available unless you deploy VoIP services. Some of the services are enhanced mobility using number portability, unified messaging, and advanced call routing. These features enhance productivity and enable access while symplifiying network infrastructure4.
The technology behind VoIP is not new, it was first implemented in 1973 at the University of Southern California by researcher Danny Cohen. The goal of the project was to develop and demonstrate the feasibility of secure, high-quality, low-bandwidth, realtime, full-duplex digital voice communication over a packet switched computer network. This first test was used to send speech between distributed sites on the ARPANET using several different voice-encoding techniques. Cooperating research companies included ISI, Lincoln Laboratory, Culler-Harrison, the Speech Communications Research Laboratory, and Bolt, Beranek and Newman.
The protocol consisted of two distinct parts: control protocols and a data transport protocol. Control protocols included relatively rudimentary "telephony" features such as indicating who wants to talk to whom; ring tones; negotiation of voice encodings; and call termination. Data messages contained vocoded speech. For each vocoding scheme a "frame" was defined as a packet containing the negotiated transmission interval of a number of digitized voice samples.
7.2.1Ease of Implementation and Enhanced Features
Tasks that may be more difficult to achieve using traditional networks are easily implemented using VoIP . For example, incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. If you take your VoIP phone with you on a trip and you connect to the Internet, you can receive incoming calls regardless of the location.
Many VoIP packages include PSTN features that most telcos (telecommunication companies) normally charge extra for, or may be unavailable from your local telco, such as 3-way calling, call forwarding, automatic redial, and caller ID.
The use of VoIP allows users to travel anywhere in the world and still make and receive phone local calls. Since the calls are immediately offloaded and onloaded to a local internet service providers at both the origination and destination points, the subscribers can avoid long distance toll charges.
Use of VoIP phones can assist in the integration with other services available over the network. Some of the services available include video conversation and message or data file exchange. These additonal services can be delivered in parallel with the conversation or audio conference.
VoIP technology still has shortcomings, but most can be mitigated using various network planning techniques. Some of the drawbacks are discussed in the following paragraphs.
A drawback to VoIP is the difficulty in sending faxes due to software and networking restraints in most home systems. However, an effort is underway to remedy this by defining an alternate IP-based solution for delivering Fax-over-IP, using the T.38 protocol or by treating the fax system as a message switching system which does not need real time data transmission. The end system can completely buffer the incoming fax data before displaying or printing the fax image.
Another drawback of VoIP service is its reliance on other separate services like a connection from an Internet Service Provider (ISP). The quality and overall reliability of the phone connection relies on the quality, reliability, and speed of the internet connection it is using. Problems with internet connection or the ISPs can affect the VoIP call. In addition, higher overall network latency can lead to reduced call quality and cause problems like echoing. This can be avoided by having service level agreements with the ISP or by using private internets.
Due to power being supplied by various power companies, VoIP will not allow calls during power outage. This is because unlike tradition phone lines power is not integrated into the VoIP network. This problem can be remedied with a battery backup systems. Another approach is to forward calls to a cell phone or alternate number during a power outage.
Due to the fact that User Data Gram Protocol (UDP) does not provide a mechanism to ensure that packets are delivered in sequential order, or provide quality of service (QoS) guarantees, VoIP implementations can have problems with latency and jitter. This problem is more true when geostationary satellite circuits are involved, which have long round trip propagation delay (400 milliseconds to 600 milliseconds). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This problem is normally solved using a jitter buffer.
While there are issues with VoIP services, most can be overcome by using proper network engineering techniques, additional protocols, or enhanced equipment.
Due to the decrease in equipment costs, network convergence, integrated services, and competition in the telecommunications evironment most traditional telcos are transtioning to VoIP services. In addition, there are nontraditional service provides like cable television and satellite television companies as well as ISPs competing to provide telephone services.
A major development, which started in 2004, has been the introduction of mass-market VoIP services over broadband Internet access services. This allows subscribers to make and receive calls as they would over the PSTN. Full service VoIP phone companies provide inbound and outbound calling services. Most offer unlimited calling throughout the U.S and often to Canada or selected countries in Europe or Asia for a flat monthly fee; less than traditional PSTN services.
According to a study by Telephia, the top ten providers in the United States include Vonage, Verizon VoiceWing, AT&T CallVantage, SunRocket, Lingo, NetZero, BroadVoice, America Online, Packet8, and Earthlink. Verizon VoiceWing and AT&T CallVantage are both listed in second place with 5.5% market share.5 The significance of this report is that Verizon and AT&T are major providers of traditional phone services. Other providers of VoIP service are companies like Skype and Net2phone that in some cases advertise free telephone calls over the Internet.
Telecommunications service providers routinely use IP telephony over a dedicated IP networks to connect switching stations. In addition they are using VoIP services to aggregate long distance taffic over data networks.
Corporations have been using IP telephony exclusively to take advantage of network convergence and toll avoidance and improve value added services to subscribers. Intel Corporation recently conducted a pilot to determine the benefits of VoIP.6 This pilot concluded and showed savings in the following areas.
Hardware Savings 55%
Add/Move Changes 52%
Data Center Savings 67%
In additon, gains were also occurred in these areas:
Significant Savings for Services (audio conferencing)
Significant decrease in Wiring Charges
Decrease in Monthly Usage Charges
The Intel case is not unique similar savings are reported by Cisco, H.J. Heinz, and numerous other companies. While savings was the initial driver, the secondary benefits of improved efficiency, productivity, and ease of use were also stressed. Finally, all companies reported that the services were transparent and the end users noticed no difference.
Existing telecommunication system are still vialble but will eventually be replaced with a new and improved system. The legacy telecommunications system should cooexist with the emerging telecommunication system. A possible approach is shown in Appendix 1.
Communication systems are currently undergoing transition to VoIP. However, if not managed correctly, implementation to support current needs may not support future requirements or systems.
The current requirements in commercial voice communications does not significantly differ from the requirements for aviation voice communications. Therefore, it is recommended that ICAO modify SARPs and develop a Technical Manual and Guidance material to support VoIP services in the ATN before the current voice communications systems become obsolete and are no longer supported by equipment vendors.
3.0 Possible VoIP Transition ApproachInitiatives in Industry
For the following discussion, the architecture shown in Figure 1 will be used as a reference.
Figure 1. Transitioning Voice Communications Architecture from TDM to VoIP
3.1 Hybrid Systems
Abrupt transitioning of legacy TDM-based voice networks to VoIP technology causes numerous problems, including:
Operational disruptions due to lack of familiarity with new equipment, capabilities, and procedures
Loss of Return On Net Assets invested in legacy equipment that has not been fully depreciated
To mitigate such issues, interim system states have been deployed wherein conversions of the infrastructure are performed in stages. Earlier stages of this process generally transition the inner core first to minimize operational impact on end systems. For ATM applications, this would affect the network core (i.e., the PSTN cloud in Figure 1), without changing the communication systems at ATM facilities.
An initial stage of a hybrid system could interface an IP-based core to the legacy end systems through an Adapter, as shown in Figure 1, which features a VoIP gateway. This level of transition may be sufficient for the user needs, or may incur costs to the limit of the budget constraints.
If there are resources allocated for further growth, latter stages of deployment would grow the IP technology outward, starting with administrative, and eventually into the operational, domains. The pace of this transition may be governed by:
Cost of IP-enabled user sets
Cost and schedule to replace legacy wiring with network-suitable cabling
Training of operational and maintenance personnel
3.2 A Protocol Approach to Hybridization: TDM over IP®
One novel approach to transitioning the network from the inside outward involves the use of the TDMoIP® protocol, patented by RAD Data Communications, which allows TDM-based end systems to interface with packet switched networks (PSN). This is achieved in two steps:
TDM traffic is adapted to capture its signaling and timing information for restoration at its destination. This is achieved by segmenting the synchronous bit stream. Mechanisms also provide recovery capabilities in case of reasonable levels of packet loss.
This adapted traffic is then encapsulated into the format prescribed by the core PSN by adding appropriate headers to the segments to form packets. These packets are then routed over the PSN to their destinations.
At the destination, the packets are stripped of their headers, the frames are concatenated, and the timing is regenerated to reconstruct the original bit stream.
An example of a TDMoIP architecture is shown in Figure 2.
7.4.1Figure 2. Example TDMoIP Architecture
With this approach, TDMoIP Circuit Emulation can run “pseudowires” of T1 or T3 traffic over IP, Ethernet, or MPLS.
TDMoIP is undergoing a standardization process in the Internet Engineering Task Force (IETF) as Internet-draft TDM over IP, draft-ietf-pwe3-tdmoip-06.txt, under the Pseudo-Wire Emulation Edge-to-Edge (PWE3) working group. In addition, TDMoIP is in conformance with ITU-T recommendations and MPLS/Frame Relay Alliance implementation agreements for TDM transport over MPLS.
3.3 Going Full Tilt: Transitioning End-to-End with Multimedia Protocols
In the advanced stages of transition, when the VoIP infrastructure is extended across the end system and core voice communication domains, the provision of services and interfacing with legacy networking may be achieved with multimedia protocols. These mechanisms offer flexibly configurable implementations that can manage multiple call streams, and have provisions for other modes of communication (e.g., video, fax).
An early entry into this arena was the ITU-T H.323 family of protocols, which features a robust and rigorous approach to multimedia management. Recently, though, the industry has been migrating towards the streamlined Session Initiation Protocol (SIP), where significant growth in application development has resulted in a broad range of capabilities and product offerings.
An important initiative that has standardized services in the telecommunications industry is Computer Supported Telephony Applications (CSTA). This serves to abstract a service layer from its underlying communications protocols. The CSTA model includes the following:
26 Call Control features
6 Call Associated features
19 Logical Device features
23 Physical Device features
5 Capability Exchange features
4 Snapshot features
3 Monitor features
17 Voice services
Various other services (e.g., Routing, Media Attachment, Maintenance, Data Collection, Accounting)
The application of CSTA to SIP is described in ECMA Technical Report TR/87, which describes the user agent CSTA (uaCSTA), which can provide much of the CSTA functionality over a SIP session.
With these tools, advanced transition stage VoIP architectures may be developed, a sample of which is shown in Figure 3. Note that the Media Servers are SIP-enabled to manage multimedia sessions, and the Media Gateway bridges between the SIP and PSTN domains (via the CLASS 5 switch).
3.4 Streamlining the End System: PBX Switching Goes Soft
Legacy PBX functionality is typically housed in dedicated bulky hardware, switching large numbers of hard-wire phone lines to achieve the various calling functions required by the ATM facility. This is a costly approach to implement and maintain, and changes in configuration are labor-intensive.
Current VoIP technology has been eliminating the hardware dependencies of voice switching with the advent of software IP PBX technology. This software is generally hosted on standard desktop computer systems. An example of such an offering is the MavianCOM 5.0, a software-based PBX, that can work over a LAN or WAN to act as an advanced VoIP phone system. With SIP as the “glue” of its architecture, capabilities are based upon the software modules “plugged” in to the framework. One of these modules that would be of interest to critical applications such as ATM is the Cascading Server Backup Module, which maintains synchronization with operational servers, keeping it ready to take over in case of primary failure. A Gateway Module is also available that routes intra-enterprise calls through the nearest, most cost-effective branch office gateway across the Internet.
Products of this ilk are becoming the norm in the VoIP industry. However, there is a growing controversial trend in the industry to adopt non-proprietary, or Open Source software, for which the source code is available. The primary example in the IP PBX domain is Asterisk. This software can run on a PC over various forms of UNIX (e.g., Linux), and on Mac OS X. This product is downloadable free of charge.
Since such software is somewhat in the public domain, it tends to be acquired economically, and is often well tested due to its transparency across the development and user communities. For those enterprises without the requisite software expertise, software integrators are available for installation and maintenance. In fact, this trend has reversed the tide in that special hardware is often provisioned to achieve optimal performance from this software! For the Asterisk software, Digium® makes the associated hardware. However, this hardware consists of interface cards that are adjuncts to existing servers, not with the big footprint, cooling requirements, and expense of “big iron” PBXs.
3.5 Hello, VoIP, are you Available?
ATM applications put a high premium on availability. For legacy equipment, this was manifested with redundancy, and specified with a quantifiable availability percentage. This worked for hardware-intensive solutions, where network paths were hard-wired and failure modes were discretely bounded.
This paradigm bears re-evaluation in the VoIP domain. For example, IP is a routing protocol, so failures in a network path or link are automatically rerouted by virtue of IP. However, VoIP systems are susceptible to more subtleties affecting availability, such as:
The flexible configurations of software and their distributed allocations to hardware make it more difficult to trace functional and performance failures
Failure modes cannot simply be described as on or off. Service degradation (e.g., noise, distortion, delay) scenarios are also to be classified as failure situations when implementing shared media network solutions such as VoIP.
Security plays a role in service availability. Since VoIP traffic shares media with data, they are vulnerable to the same threats.
The industry has responded with various VoIP management tools that are proactive and use feedback loops to monitor various functional and performance parameters to predict, alert, and sometimes prevent and correct for, imminent failures and service anomalies. There is even a tool that can provide call-blocking services if high utilization is sensed on the network. Security management tools are on the market to monitor and control user and service access to VoIP resources. Technologies have been developed for mirroring software across multiple platforms in the system, mitigating the impact of a server failure. Much of this work and other efforts to maintain high availability are being investigated under the aegis of the Service Availability Forum, a consortium of forty leading communications and computing companies. Their goal is to develop high availability and management software interface specifications for the industry.
3.6 Public Service Initiatives