Voice over Internet Protocol

Voice over Internet Protocol

Introduction:

Voice over IP is about transmitting voice signals as packets of data across a network like email. This sharing of the network introduces varying degrees of delaying of data being received that is quite unacceptable when it comes to voice calls. VoIP packets such as voice or music will have to be given priority over those carrying only data. This type of data management is only realistic if there is control over your whole network. A Virtual Private Network can be used to manage the voice priority needed and provide benefits to business and industry.

VoIP is also known as IP Telephony is a method developed to transmit packetized voice and data over packet switched networks using the Internet Protocol. The technology, voice-data convergence enables data and voice to be transmitted on a single network.

Traditionally voice is transferred using circuit-switched connections of the public switched telephone network (PSTN). This holds a connection of 64Kbps on a fixed channel between the two users on the line when a call is made and established, and will last for the duration of the call. A caller usually has to wait for someone at the other end to pick it up the handset and this introduces congestion in the network. Since phone calls involve one person talking and the other listening, there are many periods of silence during the call and hence, there is a waste of network bandwidth.

The process of VoIP starts of by first creating HDLC frames which are compressed after having been converted from a voice signal into a pulse code modulation (PCM) digital stream. The frames are integrated into voice packets where each will have a source and destination IP address. After transmitting the packets over the Internet, the routers and switches will examine the destinations associated with the packets and deliver them to a destination address. At the receiver, the packets will go through the whole process in reversed and rearranged back order according to their numbers to retrieve the voice. Packets are transmitted over a connectionless line rather than a connected line between source and destination. Breaking data into packets allows the same channel to be used among multiple users in the network and therefore The Internet is basically a connectionless network and hence VoIP can be implemented. A typical VoIP infrastructure is shown below in figure 1.

Internet Packets pass through from router to router across the network, with packet processing times at individual routers adding progressively to overall propagation delay. The transmission times impose significant transmission delays and the transmission times for individual packets will fluctuate and this is called ‘jitter’. Packet processing times will vary at a given router, and the packets associated with different voice signals can follow different routes across the network. Packets can fail to get through if the router runs out of memory to store packets and the packets can be discarded if there is no memory space left.

Figure 2: gateways facilitated a dramatic expansion in VoIP market

Evolution of VoIP

Around the mid–1990’s,advances in the multimedia capabilities of the PC, combined with low-cost Internet Access, meant that it became feasible to sustain a voice connection over the Internet. Early use of VoIP was restricted to computer hobbyists who were willing to accept the limitations. It was inconvenient because both users had to be logged on before they could talk and voice quality varied from just about acceptable to hopeless

The technology improved and VocalTec launched the first commercial VoIP client software in February 1995. The following year there was the development of gateway products that allow calls to be posted from computer to phone. This facility had a dramatic effect on the commercial importance of VoIP. This expanded the potential user base from a few tens of millions of PC users with multimedia PCs and the necessary client software to the hundreds of millions of telephone and mobile phone users. The gateway principle was subsequently adapted to enable phone to phone VoIP, figure 3.

With the development of VoIP gateways, the initial hobbyist phase of VoIP gave way to a new era, the arbitrage phase, in which the dominant commercial driver was to sell low cost telephony by using the Internet to bypass high tariff connections especially on International links. There has been an exponential growth in VoIP that is caused by the arbitrage market , figure 4.

Figure 3:

Figure 4:

The next stage in the evolution of Internet Telephony

The VoIP driven arbitrage market has been a source of competition but is insignificant when set against the hopes and expectations for the next stage of evolution of Internet Telephony. Most of the world’s public network infrastructure is either TDM based voice centric public switched telephone network (PSDN) or is part of the packet switched, data centric Internet. The next stage of Internet Telephony will be called the infrastructure phase. This will involve converging PSDN with the Internet into a single packet based network carrying voice, data and multimedia traffic, and supporting an increasing wide range of novel and innovative network services.

Cable and Wireless, a global Telecommunications group, and Nortel Networks announced in October 2000 a ten-year plan to build an International IP network.

Currently Cable and Wireless operates three distinct networks:

· A TDM based voice network, representing around 60 per cent of the company’s business

· A Frame Relay and ATM network which is the basis for a Virtual Private Network(VPN)

· An IP Network.

It is the industry’s largest VoIP development program. The C&W/Nortel deal will migrate C&W voice services into an autonomous C&W IP network covering North America, Europe and Asia Pacific.

The tree main benefits from the development is:

· Simplicity, only one network to manage instead of three.

· Better bandwidth use on a voice call, TDM wastes at least 50 per cent of the bandwidth. The bandwidth gain will vary with traffic type but overall less bandwidth will be needed.

· Flexibility, virtual private networks will be simpler to configure and novel network services will be easier to deploy.

Congestion Avoidance

The secret of congestion avoidance is adequate bandwidth combined with control over access to bandwidth. The C&W network will thus be end to end ‘private IP cloud’, controlled solely by C&W. Within the IP cloud, network performance is further enhanced by use of resource reservation protocol (RSVP) and multi-protocol label switching (MPLS). RSVP is a mechanism for defining paths across the network that can meet the bandwidth requirements for traffic such as voice and multimedia with a specific time requirement.

Conventionally, routing decisions are made on a router to router basis, which can result in congestion on some links and under utilization on other links. MPLS simplifies and hence speeds the routing function at individual routers. Routing can still be managed at the network level by MPLS having the ability to define a complete route across the network, which is the labeled-switched path (LSP). The route will be specific to a given source and destination pair, and can also be varied according to traffic type. In an instance, the LSP can be defined by an appropriate QoS scheme, such as RSVP.

Bandwidth management is only part of providing transparent connectivity with PSDN. It is necessary to address the fundamental differences in transmission management and signaling between the PSDN and the Internet domains. Under the basic Internet Protocol, each packet is conveyed independently, there is no’ session’ or ‘state’, corresponding to a PSDN connection. The IP world does not understand SS7, the PSDN signaling system used to implement such services as calling line identification.

The C&W network uses two basic building blocks to overcome these problems:

Gateways that pass traffic between PSDN and the IP core, and Softswitches, essentially computer servers, that run the software which controls the gateways in order to enable IP sessions that mirror conventional PSDN connections.

The network core uses MPLS enabled Cisco routers, while the gateways and softswitches are being supplied by Nortel Networks. Further technical developments from these products are required to achieve the ‘necessary level of performance’, but the network is expected to be ready for pilot trials towards the end of 2002, and with the mass market taking off at the beginning of 2003.

In parallel with the migration effort, C&W is implementing a more near term offering for new VoIP services such as VoIPVPN, H.323 or SIP phones, and IP PBXs using the same MPLS enabled Cisco routers connected to Cisco gateways controlled by a softswitch.

Worldcom, an international long distance carrier is approaching the concept of VoIP by supplying VPN voice services using TDM. Since TDM is cheap and cost effective, the transition to VoIP will be subject to service benefits. PSDN evolved as a single application voice system, which involved signaling closely coupled with the telephone circuits. The physical integration and development of new services will be difficult and time consuming.

With IP, networks are not physically connected to the control and transport infrastructure. To introduce new services, appropriate code for a control server can be written, and the server can be located anywhere on the network.

There are agreed protocols to enable servers to communicate with the networks ‘ingress and egress’ gateways, and establishing and terminating service sessions over the network.

Two protocols currently used for this purpose are the ITU’s H.323 system and the SIP system.

H.323 is a standard for providing a foundation for audio, video, and data communications across IP-based networks, including the Internet. By complying with H.323, multimedia products and applications from multiple vendors can interoperate, allowing users to communicate without concern for compatibility. The H.323 standard defines a group of standards for use with multimedia communications over local area networks (LANs) that do not provide a guaranteed QoS. It was originally designed for use with a single LAN, modifications have been introduced to make it suitable for WANs and enable wide spread communications.

SIP means session initiation protocol was developed under the Internet Engineering Task Force. SIP is the preferred method because it is easier to use and is modeled on http, hypertext transfer protocol. SIP is also part of Microsoft’s XP operating system, which will expand SIP applications.

Worldcom has linked SIP based control servers to its public Internet network, UUNET, and is marketed under ‘IP Communications’. Worldcom is offering customers overall communications over IP, separate voice and data. This will be multimedia, voice, video, fax, messaging and data combined on a single IP VPN. The benefits of ample bandwidth are augmented by the use of MPLS in the network and prioritization techniques speed up the transfer of voice and video.

IP communications was launched in the US in 2001, and is expanding into Europe, Africa and Asia in 2002.

The Office

Office workers can access access telephone services by ‘logging on’ to the network. Home Workers can access the facilities of the office phone system if they have broadband connections. Thus VoIP is applicable on both WAN and LAN.

The need for separate telephony cabling system is illiminated by using the IP telephone, that offer the functionality of a digital PABX handset but connect directly into the LAN as opposed to having to be connected to a PABX.

The IP telephone can be configured and installed easily like a PC and can be plugged in anywhere on the LAN or WAN. An IP PABX is required and can support both IP telephones and conventional telephones which is a hybrid solution. This can help the gradual transition from PSDN to IP. A line card for every office telephone is no longer required and the industry standard computing platforms means that IP PABXs are more cost effective than other PABX products. In the meantime Network Switches can be installed to make networks ‘VoIP ready’ so as not to have to replace existing conventional PABXs and meet the QoS requirements as required for VoIP.

The Technology promises substantially cheaper voice and fax services through the use of public and private infrastructure for trunk connection of calls. Voice connection could be between two PCs or IP Telephones that are connected to the IP network. The terminals must have the required software to convert the analog voice to digital IP packets with source and destination address.

A connection must be made between the terminal and the traditional public telephone or a connection between two telephones going through the IP network. This is the function of an IP telephony gatekeeper and gateway, the most important elements of VoIP technology.

VoIP and Internet Telephony

Current telephone technology uses PSDN or Public Switched Telephone Network. A dial tone from a Class five switch occurs when the Telephone is lifted. This equipment provides call waiting, caller ID and orther features. Local calls stay on a Class 5 switch but national calls move over to a Class 4 switch. VoIP technology replaces Class switching technology to a system that is efficient and inexpensive.

VoIP uses Internet Protocol Technology that converts voice to digital packets and moves the packets along via routers and gateways. The standard telephone can be connected to place a call through the Internet Service Provider, the ISP. The process is transparent to the caller and the called party.

Internet telephony is different to VoIP because it uses data-handling capabilities of the Internet to carry voice over computer networks. The vocal quality is not good because the Internet does not handle voice efficiently like data. The computer’s microphone and speakers are used to interface with the computer Internet telephony software package. Voice content is processed in data streams resulting in an inferior quality to that of VoIP.

Comparing VoIP to Internet Telephony VoIP:

· is more flexible

· is based on open architecture that provides more services.

· uses Internet Protocol via routers and gateways that handle voice and data separately.

Internet Telephony:

· treats voice the same way as data · provides low-level voice service

· does not handle latency well so some voice data is dropped along the way

IP telephony has not been an instant success in the UK and Ireland, partly because of the cost of Internet access, and partly because a fairly powerful computer with a good quality sound card and a reasonably fast connection is needed to make it work. The quality is more like a CB radio than a landline when first experimented upon in 1995. A permanent broadband Internet connection is required to make VoIP economical and efficient as well as convenient. NEC are including Callserve software with their computers and VoIP looks set to go into the mainstream of future technology.

Early VoIP was limited to having to be connected to a server and talking to people on the internet, such as in ‘chat-rooms’. In the future calls from PC to PC could be made and calls direct from a PC to any telephone in the world.

Challenges facing VoIP

The main problems will be delay and jitter. The maximum tolerable delay for voice is 400 ms, and C&W are aiming for an upper limit of 250ms on international calls, while calls within Ireland and the UK should incur a delay of no more than 150ms. The maximum jitter on any call should be no more than 1ms and the packet loss should be less than 0.1per cent. C&W will achieve these results by avoiding congestion and augmenting the standard Internet protocol stack to speed the routing process and deliver a Quality of Service.

For future development, there are certain issues involving VoIP that need to be concerned before it becomes a viable means for providing voice communications.

Delay in voice communications comes from the process of voice packetization, transmission, decompress and reassembling them again in correct order. It can cause two more problems: echo and overlapping. Echo is caused by the caller’s voice, which is in signal form, being reflected back from the far-end telephone equipment back to the caller. This can cause serious quality problem and the VoIP system must overcome this by means of echo control & echo cancellation mechanisms. Overlapping happens when a person's speech is overlapping with the other talker's speech during conversation. End-to-end delay must be reduced by the VoIP system to remedy this. Currently, delay can be minimized by using several solutions including Real Time Protocol (RTP) where voice packets will have priority over data packets in a network system and IP Packet Segmentation which prevents long data packets from delaying a voice packet during transmission.

Jitter happens due to the variety of inter-packet timing which in turn, originates from the network system. While transfering packets through the network, a packet might transverse and cause jitter to the conversations. If the effect is highly significant, it can distort the natural rhythm of speech patterns, causing another bad quality of voice communications. One solution for removing jitter is to collect the packets and hold them long enough to avoid any packet from transversing. The slowest packet must be allowed to arrive before rearranging the packets into the correct sequence. This can cause additional delay and hence removing jitter must be compromised with minimizing delay.

Packet Loss due to the nature of IP networks is no guarantee that packets will be delivered in a certain order. Under peak loads and heavy congestion period, packets might be dropped in the middle of transmission and loss. To overcome this, some Internet broadband providers such as Quest and Level3 are starting to provide their own VoIP services to the public. The excess bandwidth that they have enables them to offer digital circuit switched quality calls using the IP protocol.

VoIP Architecture:

A typical VoIP network includes the following components: · Media Gateways

· Signaling Gateways · Gatekeepers · Class 5 switches · SS7 network

· Network Management System · Billing Systems

Testing of VoIP

Testing of VoIP is a tri-fold task, functionality, standard compliance and performance verification are necessary.

A successful pre-deployment testing procedure must address · functionality

· functionality under stress · fault insertion tests

· all functions work properly and consistently via long term stability testing · verify performance versus compliance with system requirements

Changes in software and hardware version upgrades can cause degradation in functionality, quality and performance.

Testing ideally is done in a lab environment so as to minimize deployment, troubleshooting, operational and maintenance costs. When functionality tests fail, a detailed protocol implementation is required to verify the conformance of VoIP devices.

This requires a detailed decoding capability of VoIP all protocols. H.323 use the ANS.1 notation while SIP and Megaco use ASCII messages.

Effective pre-deployment testing follows a well defined methodology that address the variety of issues that can impact the networks adherence to specifications to a real world environment.

Consideration for number of anticipated users and estimated traffic per user. As well as network infrastructure such as Frame Relay, ATM, and DSL.

The Poison statistical model, predicts end user behavior, the average call is 180secs, the VoIP network specifications can be defined using the parameters,

· blocking, · Busy hour traffic · Centi-call seconds · Erlang

A gateway converts a telephone conversation into the correct format as data packets to enable it to travel across a data network as Internet Telephony. Gateways are necessary at both ends of a telephone conversation so that voice can be converted and reconverted back into an intelligible language at the other end. VegaStream are a company that provides products and Internet telephony solutions. They have the Vega 50, which has eight ports to allow eight simultaneous conversations. The Vega 100, the Vega 200 and the Vega 5000 are also available.

Gateway testing involves a variety of aspects

· Compression and decompression · Bandwidth utilization

· Jitter

· Echo cancellation · Alternative re-routing · Fall-back to PSTN

The gatekeeper is the traffic controller of VoIP. It determines the call routing scheme and its correct operation under stressful network conditions. This is important so as to provide an adequate grade of service.

IVR (Interactive Voice Response) is an integral part of a business phone system. It reduces operational and human resource costs. IVR systems use DTMF (Dual Tone Multi-frequency) tones to transfer user requests to the system. DTMF tones are the same tones used for tone dialing and are the sums of two sine wave tones.

Overview of Cisco VoIP Infrastructure Solution for SIP

The Cisco VoIP Infrastructure Solution for SIP implements a voice-over-packet network design using SIP to provide telephony services. It lays the foundation for building SIP-based VoIP solution, which is build using Cisco products. Components used implement toll by-pass, effect Dedicated Access Line (DAL) replacement and provide enhanced IP telephony services such as a scaleable private number plan, and to provide desktop services such as call forwarding, call hold, and call transfer.

The solution includes:

· a SIP telephone · a SIP gateway · a SIP proxy server

· a unified messaging server · a firewall

· a VoIP solution for service providers

These components work together to provide a SIP-based VIP solution that can be integrated with existing telephone systems

There is a phased implementation procedure of the Cisco infrastructure solution to VoIP and the solution is implemented from an intranetwork and an internetwork approach.

The intranet-work “phased” implementation of the Cisco VoIP involves first replacing the traditional Dedicated Access Line (DAL) and by-passing carrier toll lines by introducing Cisco SIP gateways and an IP network between the private branch exchanges (PBXs) as follows:

The next phase involves introducing SIP proxy servers to provide support for a scalable private number plan to the IP network as follows:

The next phase is the addition of Cisco telephones, which connect directly to the IP network. They provide features such as Call Waiting, Call Holding, Call transfer and Call forwarding.

A Radius Server is integrated into the IP network with the SIP proxy servers to provide application services. This enables the SIP proxy servers to perform Authentication. It also provides end customers with enhanced services, such as ‘find me’ and ‘call screening’. The Sip gateways interface with the application services using the Radius server for billing purposes.

A unified messaging server is added to provide voice mail

Summary:

· A Quality of Service IP Network is central to the network using Cisco internetworking equipment with a set of Cisco Gateways and one or two proxy servers.

· The Cisco SIP gateways are connected to the PBXs using T1 or E1 lines with channel associated signaling (CAS) or primary rate (PRI) signaling.

· Telephones and Fax machines are connected to the PBXs · SIP IP phones are connected directly to the IP network

· A server running a unified messaging service is connected to the IP network

· Sip is used for signaling between Sip clients, the Cisco SIP IP phones , the Cisco SIP gateways, and the SIP proxy servers

· RTP/RTCP is used to transmit voice data between the SIP endpoints after sessions are established.

The Internet-work phased approach is as follows, this is the solution to integrating a SIP enabled VoIP network with the public network (PSDN) structure

PIX firewalls are added to provide inside security

Cisco SS7 Interconnect for Voice Gateways Solution components are added so as to integrate the SIP enabled VoIP network with a public infrastructure

When calls are made within a single SIP IP telephone network, the process involves the origination and destination phones and a single proxy server

1. Phone A initiates the call by sending an INVITE message to the SIP proxy server

2. The SIP proxy server interacts with the location server to determine user addressing, location and features. It may also interact with application services.

3. The SIP proxy server then proxies the INVITE message to the destination phone.

4. After a response and acknowledgement are exchange with each phone, an RTP session is established between Phone A and B

To process calls between SIP IP telephony networks, the process involves the origination and destination phones as well as two or more proxy servers

1. Sip Phone A initiates a call by sending an INVITE to the SIP proxy server

2. The SIP phone can obtain additional information from the Radius server to interact with application services.

3. The SIP proxy server in phone A’s network contacts the SIP proxy server in phone B’s network

4. The local proxy uses the domain name system (DNS) to determine if it should handle the call or route to another proxy server.

5. The SIP proxy server in phone B’s network may interact with application services to obtain additional information.

6. The SIP proxy server in phone B’s network then contacts the destination phone.

7. After responses and acknowledgments are exchanged, an RTP session is established between the Cisco SIP phones A and B.

When calls are made between a SIP IP telephony network and a traditional telephone network, the process involves the originating phone, one or more proxy servers, a gateway, and a PBX or PSDN device

1. The Cisco Phone sends an INVITE to the SIP proxy server

2. The SIP proxy server might interact with RADIUS for application services 3. The SIP proxy server proxies the INVITE to the Cisco gateway

4. The Cisco gateway establishes communication with the traditional telephony network 5. An RTP session is established after responses and acknowledgements re exchanged.

Altera and Xilinx are among companies investing towards the development of VoIP using their Field Programmable Gate Array technology.

Spartan-11 FPGA family was introduced by Xilinx as solution for various VoIP products implantation. The Implementation of Spartan 11 is shown below

Spartan-11 is chosen for VoIP products due to its feature of providing large number of gates for a low price and flexibility.

Altera Corporation introduces its FPGA products called Apex 11 for the same purpose.

The gateway network system consists of a shelf which contains PSDN cards, 2 digital access cards and 3 bus lines known as time division multiplexing (TDM) bus, control bus and packet bus. The PSDN cards interface between the telephone hardware and the routing network while the digital access card (DAC) interfaces with the Internet infrastructure.

The TDM bus delivers unpacked voice data between all the cards, the packet bus

moves voice traffic in packets between the DAC and the Control bus and manages the communication between the DAC and PSDN cards.

The advantage of FPGA is that functionality is implemented inside one APEX 11 device instead of being implemented by different processors, as is the case with PSDN. Another advantage is reduced cost by reducing the number of processors needed inside the card.

These devices from Altera and Xilinx are just some of the examples in which FPGAs are being used as better substitutes for ASICs in VoIP applications. Other FPGAs such as the Atmel Smart Internet Appliance Processor (SIAP™) from Atmel Corporation are designed to be implemented in VoIP and Internet audio appliances.

The DAC main purpose is to prepare data to be transferred over the Internet. It unpacks the packets received from the Internet and converts it back to analog voice used by the shelf. The card also contains an APEX II device. The APEX module converts voice to HDLC packets does echo cancellation, control and processing functions.

Advantages of FPGA

1. Flexibility and can be Reconfigured

The forward and reverse process of packetizing the voice requires several processors and devices to implement each and every function. Using an FPGA will allow the functions to be implemented inside one reconfigurable processor where it will reconfigure itself depending on the specific function at a specific time. This special feature reduces the costs needed to develop various VoIP products. New features can be implemented without sacrificing functionality. Users can remotely configure upgrades as advanced features are released

2) Time to market

In the expanding and rapid growing of the telecommunications market nowadays, time for a device to enter the market is crucially important since the competition of providing new technology means that the product that enters the market first and wins the market. Since FPGAs can easily be programmed and debugged without having to wait for the development time as with ASIC, it can be marketed faster. If there is a challenge of a new product, the FPGA can easily be upgraded without having to wait for a development process. This means the life-span of the FPGA device can be longer than that of an ASIC as the core processor.

Conclusion

VoIP has advantages over PSDN because it optimizes the transmission bandwidth so that voice, data and video can be transmitted simultaneously in packet format over a network. The flat rate pricing of the Internet together with the low cost implementation of VoIP due to the single network data-voice convergence results in much cheaper way of data, voice and video communications around the world.

Voice-over-Internet-Protocol technology has been expanding and growing rapidly in recent years together with the Internet and networking technology. It is still in the state of improving and developing into a better technology from day to day. It involves a complex and a large number of operations from the first step of converting the voice to PCM format and forming HDLC packets to recovering and reassembling the voice at the destination. The future of telecommunications will depend on engineers solving problems such as delays, jitters and packet loss. Engineers must provide a quality of service to the public so as to implement VoIP as a viable communications technology.

IP networks offer ways for businesses to reduce costs and add meaningful capabilities. But for each organization, the starting points and quality assumptions will always be the existing telephone and networks that are in place. Where the goal is to introduce new applications, design engineers can move more aggressively into VoIP as long as they design the networks with sufficient bandwidth to ensure proper voice performance.

VoIP solutions need to be implemented as transparently as possible into existing systems, while achieving cost savings and establishing the platform for new applications. As networks get larger and maintenance costs continue to rise, a VOIP network for voice communication becomes more attractive. Because VOIP is a major step in a new direction for voice communication, many solutions have been implemented in an attempt to best serve the marketplace. Standardising these solutions will lead to the presence of a global VoIP implementation resulting in services that are cheaper, advanced, and have redundancy features. VoIP technology will revolutionize the way voice traffic is delivered.

References and links:

Members Magazine of ‘The Institute of Electrical; Engineers’, January 2002: The Convergent Phone

Voice over Internet Protocol (VoIP) By: Mark Kates, Derek Matthewson

http://www3.sympatico.ca/mkates/portfolio/voip/VoIP.htm

Overview of the Cisco VoIP Infrastructure Solution for SIP

http://www.cisco.com

FPGA implementation on Voice-over-Internet Protocol

by Ahmad Affzan Abdullah under the supervision of Prof. Ian Page

http://www.nfoeng.ee.ic.ac.uk http://www.iptelephony.org

Voice over IP Testing – A Practical Guide, Radcom White Paper, www.radcom-inc.com http://www.protocols.com