1. Abatract
The Internet is being modified to support voice traffic and products are being made to link the data and voice networks. Eventually the Internet and the telephone network will be one and the same.
Internet Telephony is an emerging technology and has a number of technological and evolutionary issues. The technological issues are mainly because the Internet was not designed for real time traffic such as voice. The evolutionary issue is the fact that a variety of vendors develop their products according to market demands and supplies. It will take time for all these products to converge and inter work with the same reliability as the circuit switched networks.
VoIP, VoFR and ATM are the subjects of the technology nowadays. Each of them has advantages over the others, and for the benefice of the technology they all need to be interwork.
The following article describes the tree technologies, VoIP, VoFR and ATM by an overview of each of them, the interoperability of them and the Future potential of these three technologies
2. Introduction
VOIP is growing fast. The very knowledge of the applications of this technology is enough for users and manufacturers to flock towards it. It is ideal for computer based communications and at the same time bringing down the cost of multimedia transfer. Hence VOIP products and services have flooded the market.
When Frame Relay technology was designed, it was without the option of carrying voice. The engineers have worked hard under the need of the vendors to make the voice together over the frame relay and give a good quality of voice in parallel of the high speed of the FR.
ATM was designed to be a multimedia, multi service technology. Though the ATM has been accepted by the marketplace for its ability to deliver high speed data services, till the recent past its potential for deploying for voice services was overlooked. With the competitiveness of today's market, the network operators and the service providers have been continuously striving to reduce operating costs and lift network efficiency. They recognized that significant economic benefits can be achieved once the data traffic and voice traffic are integrated onto a single network. Since ATM has been around for around a decade claiming to be a multimedia technology, most of the service providers have started installing
single ATM infrastructure to support voice, video, and data transfer.
Network engineering still working and developing the packet networks (Frame Relay, IP and ATM) to carry voice as well as data, and they are searching to reassemble these technologies into a single communication service, to develop interconnection and internetworking standards in order to deliver voice services over Frame Relay, IP and ATM.
3. Overview of VoIP
Internet Voice, also known as Voice over Internet Protocol (Voice over IP), is a technology that allows you to make telephone calls using a broadband Internet connection instead of a regular phone line. Some services using Voice over IP may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some services only work over your computer or a special Voice over IP phone, other services allow you to use a traditional phone through an adaptor.
Voice over IP allows us to make telephone calls using a computer network, over a data network like the Internet. Voice over IP converts the voice signal from our telephone into a digital signal that travels over the internet then converts it back at the other end so we can speak to anyone with a regular phone number. When placing a Voice over IP call using a phone with an adapter, we'll hear a dial tone and dial just as we always have. Voice over IP may also allow us to make a call directly from a computer using a conventional telephone or a microphone.
Voice over IP lets us make long distance voice and fax calls over existing IP data networks instead of the public switched telephone network (PSTN). Today businesses that implement their own Voice over IP solution can dramatically cut long distance costs between two or more locations.
3.1 Functionality
VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:
Incoming phone calls can be automatically routed to our VoIP phone, irrespective of where we are connected to the network. Take
the VoIP phone with us on a trip, and anywhere we connect it to the Internet, we can receive our incoming calls.
Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection.
VoIP phones can integrate with other services available over the Internet, including video conversation, message or data file exchange.
3.2 Implementation
Because IP does not provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, VoIP implementations may face problems dealing with latency. They are faced with the problem of restructuring streams of received IP packets, which can come in any order and have packets delayed or missing, to ensure that the ensuing audio stream maintains a proper time consistency.
Another main challenge is routing VoIP traffic to traverse certain firewalls and NAT. Intermediary devices called Session Border Controllers (SBC) are often used to achieve this, though some proprietary systems such as Skype traverse firewall and NAT without a SBC by using users' computers as super node servers to route other people's calls.
Keeping packet latency acceptable can also be a problem, due to network routing time and transmission distances.
3.3 Technical details
There is a lot of debate about the two most popular types of VoIP; SIP and H.323, each of them has its own merits, H.323, was the most popular protocol, though its popularity has decreased in the "local loop" due to its poor traversal of NAT and firewalls. For this reason as domestic VoIP services have been developed, SIP has been far more widely adopted. However in backbone voice networks where everything is under the control of the network operator or Telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being terminated over VoIP. So really SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone". With the most recent changes introduced for H.323, however, it is now possible for H.323 devices to easily and consistently traverses NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously.
Where VoIP travels through multiple providers Soft Switches the concept of Full Media Proxy and signaling proxy are important. In H.323 the data is made up of 3 streams of data: 1) H.225.0 Call Signaling 2) H.245 3) Media. So if we are in London, our provider is in Australia, and we wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600ms) and packet loss will be high. However in signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. these proxy concepts could lead the way to true global providers.
One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically to send a G.723.1 5.6kbps compressed audio path will require 18kbps of bandwidth based on standard sampling rates. The difference between the 5.6kbps and 18kbps is packet headers. There are a number of bandwidth optimization techniques used such as silence suppression and header compression this can typically save 35% on bandwidth used. But the really interesting technology comes from VoIP off shoots such as TDMoIP which take advantage of the concept of bundling conversations that are heading to the same destination and wrapping them up inside the same packets. These can offer near toll quality audio in a 6-7kbps data stream.
3.4 DSL Internet access
VoIP technology does not necessarily require broadband Internet access, but this usually supports better quality of service. A sizable percentage of homes today are connected to the Internet through DSL, which requires a traditional phone line. Having to pay for VoIP in addition to both a basic phone line and broadband Internet access reduces the potential benefits of VoIP. However, some regional telephone companies now offer DSL service without the phone, thus saving us money when we switch to VoIP. VoIP can also be used with Cable Internet instead of DSL, eliminating the need to purchase two telephone lines.
3.5 Reliability
Conventional telephones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages. In order to use VoIP during a power outage, an uninterruptible power supply or a generator must be installed on the premises. Early adopters of VoIP
may also be users of other phone equipment, such as PBX and cordless phone bases that rely on power not provided by the telephone company.
3.6 Security
The majority of consumer VoIP solutions do not support encryption. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. There are several open source solutions like VoIPong or Vomit that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications; however such security by obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are usually not available at a consumer level.
3.7 Benefits of the technology
The integration of voice and data traffic will be demanded by multi application software.
An integrated infra structure that supports all forms of communication allows more standardization and lesser equipment management.
The integration of voice and data effectively fills up the data communication channels efficiently, thus providing bandwidth consolidation. The idea is to move away from the TDM scheme wherein the user is given bandwidth when he is not talking. Data networks do not do this. It is a big saving when one considers the statistics that 50% of a conversation is silence. The network efficiency can be further boosted, by removing the redundancy in certain speech patterns.
In general, phone service via VoIP costs less than equivalent service from traditional sources but similar to alternative Public Switched Telephone Network (PSTN) service providers. Some cost savings are due to using a single network to carry voice and data, especially where users have existing under-utilized network capacity they can use for VoIP at no additional cost
4. Overview of VoFR
Frame relay is packet switched network that was designed for transmitting data over fixed line.
The frame relay can be a long distance telephone service. Once the service is established, the customer only need to transmit his data over a local link to a nearby frame relay station so the frame relay is responsible for transmitting the data of the user and delivering to destination. A frame relay service provides many attractive alternatives to leased lines.
4.1 Characteristics of a frame relay
One of the first noticeable network is its very high transfer speeds. The data transfer speed can be very fast as same as network.
Frame relay network also provide very good security. Because of the encryption technique used to transmit data between frame relay switches, also the frame relay connection are permanent then it is available.
Frame relay was originally designed and used to transfer packets of data between two sites more cost effectively than leased lines could.
Voice over frame relay allows the internal telephone systems of companies to be connected using frame relay PVCs.
4.2 Advantages
Transferring telephone calls using frame relay has a number of advantages over using the leased line service of a standard telephone system.
Frame relay reduce the cost of a telephone call.
Frame relay uses network resources more efficiently by combining a number of channels of voice traffic with data and reliably transmitting the result over an existing frame relay network.
The cost of the equipments that connects a company’s PBX to the frame relay network is quickly recovered from the saving involved in avoiding conventional telephone line. Because VoFR does not significantly complicate network architecture or increase link speed, voice, fax and data traffic can be combined effectively over a single network of wires.
Up to 255 voices sub channels can be multiplexed on a single frame relay circuit.
4.3 Disadvantages
Data network called upon to transmit frame relay voice in addition to data, it may experience congestion problems. Many corporate
networks are already straining just to deliver data. To add voice to an already congested network is a problem.
Voice compression is necessary in frame relay application to help ensure high quality audio while maximizing bandwidth usage. Sometimes this voice compression can affect the quality of the signal and therefore the sound quality of the voice on the telephone line.
5. Overview of ATM
Asynchronous Transfer Mode ATM, is a multi service, high speed, scalable technology. It is a dominant switching supporting services with different transfer characteristics. ATM transports voice, data, graphics and video simultaneously at very high speeds,
ATM can be used as the transfer technology for local area networks, metropolitan area networks, and wide area networks, since ATM like frame relay, is a layer 2 protocol, it can be supported by many different types of physical layer media, such as twisted pair and fabric optic cable.
ATM can support different classes of traffic to provide different levels of service (QoS).
5.1 Advantages of ATM
With the standards are in place it is now possible for packet switching techniques like Frame Relay or ATM to deliver high quality speech. Some of the intrinsic advantages ATM has over other network technologies are listed below.
The very format of the cell was arrived at by considering data, voice, and video payload requirements. ATM cells are of fixed size, 53 bytes each with 48 bytes for payload and 5 for ATM header. This helps in reducing the packetization delay significantly, which is one of the major delay parameters.
It supports extensive QoS (Quality of Service), which allows voice traffic to be transmitted across the network in a reliable jitter-free way.
Various service classes’ capabilities are supported by various ATM Adaptation Layers (AALs).
ATM switches have always been designed with effective traffic management capabilities - for example, call admission control, usage parameter control, traffic shaping, etc.
Single network for voice, data, and video.
Interworking with PSTN is relatively straightforward.
5.2 Disadvantages of ATM
It is often more expensive than other data transmission options. The cost of ATM equipment is high; Due to the complexity of ATM there is a high learning curve for setting and managing the network. Compatible hardware and software may not be widely available.
ATM is now being heavily used by the large telecommunications carriers to provide voice and Internet services. Because of its complexity and cost, smaller businesses have been reluctant to use ATM.
5.3 Standards and Specifications.
Various applications are available for the transport of voice over an ATM network. Each application has differing requirements for voice transport based on what class of network operators they are defined in. The three major classes of network operators are:
National or International Operators typically have an extensive PSTN service operating over SDH/SONET or PDH infrastructure. When bandwidth is limited there will be a requirement to integrate voice and data traffic, for reasons of efficiency into a single ATM network. Within the local loop, ATM may be a valuable solution for the carriage of voice and data to business premises.
Alternate Carriers or Value Added Network Suppliers take up licenses to provide communication services in competition with the incumbent national operators. Instead of having their own transmission infrastructure, they buy bandwidth from the primary operator. Cost and limited availability of bandwidth demand ATM's efficiency, and integration of voice and data services. Example of alternate carriers is cellular phone operators.
Private/Enterprise networks buy bandwidth at commercial (retail) rates and achieve the most they can with the resources on hand. Such organizations will have already deployed a TDM network utilizing E1 or T1 links. They will be looking to integrate these solutions into a new ATM network, and gain improvements in network performance and efficiency.
6. Data and its implications over voice
IP, ATM and Frame Relay are not as same as normal telephone, public telephone carries voice others carry data and cells known as packet and cells switching.
In circuit switching network (telephones) once the transmission start the line is occupied even though the line is in hold which means that the third party can not use
the path in any way, this is not efficient recording to packet switching.
One of the main reasons that this network (packet switching) is reliable because it allocates the bandwidth to the links, and each link can use the bandwidth available at any time.
One of the problems packet switching face is the delay, because of the share of the line, as a result of congestion, and in other situation could be the drop of packets so the delay must be minimised as minimum as possible to get a better quality of voice, and the integrity of the transmission.
Data and voice are tow distinct things, the voice is very sensible and any small delay will affect directly the quality of the voice, the voice will be not understandable in most cases.
One solution can be taken to solve these problems of congestions and delays is to use a bandwidth to the network in a way that will be not increased but managed by a mechanism that make priorities and reduce the congestions and delays.
7. The interoperability of the three technologies
Operating data and voice together with IP,ATM and Frame Relay is the best network solution of any other technology, however, the standardization of the elements such as protocols, voice compression and other elements are not compatible and then not standard to inter-work between them. Standardization is very important for the three technologies, and without it no mean to the interoperability.
VoIP interoperability defined by ITU H.323 without the definition of the address encoding and security and other things which is mean that the definition is not complete, based on these kind of definitions it will be no interoperability between equipment of different vendors.
IP vendors are collaborating and working together to achieve interoperability gatekeepers and gateways of different vendors to deploy different IP platform at either end of the network.
Frame Relay is a Permanent Virtual Connection (PVC), the reason why the RF.11 Phase 2 is not yet implemented, the two ends use
the same vendors equipments, in the meantime vendors are trying to interoperate the equipments for interoperability that the market requires.
With Voice Over ATM, the problem of the interoperability is there, AAL2 is not yet standardized, therefore the problem of interoperability between vendors still there, as same as FR and IP.
8. Future potential of these three technologies
With more and more voice traffic moving onto data networks. Vendors of voice equipment will continue to develop integrated voice and data devices based on pocketsize technology.
RAD Data Communications will remain in the forefront of industry efforts to provide universal services, for the benefit of the users
Since standardization has not been adopted for any one technology, it is not possible to expect the interoperability standards between technologies in the near future. It is essential that the interoperability be transparent to the users, who want to communicate through the network efficiently and without concern for the technological issues involved.
Due to the lack of interoperability standards for voice communications over Frame Relay, IP and ATM, vendors must develop proprietary interworking solutions.
RAD is developing a pre-standards strategy to facilitate interworking between Frame Relay and IP. The strategy will provide a migration path from Frame Relay to IP technologies, which can be an important advantage when IP services become available. RAD is currently developing an interworking solution between Frame Relay and IP. The VoFR-VoIP product will perform signalling conversion and negotiate with the remote IP product in order to choose a common voice compression algorithm and other parameters.
9. Conclusion
The integration of voice and data networks has been a long time coming, and complete unification is still a distant goal for most organizations. Consolidation is progressing in parts of the network, especially where medium-term cost savings can justify a new approach. Each of the major packet-voice-transport options--IP, frame relay and ATM--has its strengths, and no single technology is the best fit for every situation.
ATM is mature, reliable and manageable. However, its limited span and high cost per port will discourage IT managers from deploying it end to end. Inexpensive and fairly reliable, voice over frame relay is
a good interim step, but best used by only a few hundred sites per backbone. And voice over IP, with its wide reach, shows great promise but has some hefty requirements for maintaining its quality.
Although hundreds of vendors are exploring these technologies, four have taken the lead. Cisco, Lucent, Nortel and 3Com.
Interoperability between the various networks will allow users to benefit from the best that each network has to offer. The extent of compatibility is limited by the prioritization methods and signalling protocols, even though these networks follow similar fragmenting techniques. The level of interoperability will increase with the introduction of standardizations within the protocols, which will facilitate the interworking.
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10. References:- http://www.fore.com/products/wp/voicewp.htm
- http://www.cisco.com/warp/public/cc/cisco/mkt/switch/cat/8500/index.shtml.
- http://www.fore.com/products/voice-plus/voice-plus_po.html
- http://en.wikipedia.org/wiki/Special:Search?search=voice+over+ip&fulltext=Search
- http://www.freedomcomms.com/newcontactform.asp
- http://www.protocols.com/papers/voe.htm
- http://www.rapid.co.uk/iptelephony.html
- http://www.telsyte.com.au/feature/voip.htm
