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Issue of August 2002 
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VoIP's come a long way

Since its introduction in 1995 VoIP technology has come a long way. Enterprises in India and the rest of the world have successfully used the technology and derived cost savings and other benefits from it. A look at the technology in its current evolved state can help you plan ways to use it optimally in your network. by Soutiman Das Gupta

VoIP was introduced way back in 1995 and since then has come a long way. It has now been adopted universally as a reliable technology alternative by which voice is transmitted over a data network instead of a traditional voice network. Companies who have successfully deployed the technology have realized numerous benefits.

The biggest benefit has been savings in communication cost. By using IP Phones to make long distance calls between branch offices and zonal offices, enterprises effectively spend a fraction of the amount it would spend if it went the traditional telecom network way. The next big benefit has been optimum utilization of network resources. Many enterprises have built robust networks with updated routers, switches, and other devices, and put high bandwidth links in place. In many cases, a lot of bandwidth and other resources were available for use even after critical enterprise data traffic was transmitted. Transmitting voice through this unused facility makes better use of the network resources.

Since our government relaxed certain restrictions on the use of VoIP from April 2002, many Indian enterprises have successfully leveraged the technology to their benefit. Even before this date, many enterprises implemented VoIP in a closed user group, as was permitted earlier.

The effective use of QoS (Quality of Service) in the network devices has helped overcome limitations like delay, echo, and jitter. Here's a look at the technology, the protocols, and the QoS factors.

Understanding QoS

Generally there are three techniques that can be used (separately or in combination) to improve network QoS.

Controlling the networking environment: You have to provide a controlled networking environment in which the capacity can be pre-planned and adequate performance can be assumed.

Using management tools: You can use management tools to configure the network nodes, monitor performance, and manage capacity. Traffic can be prioritized by location, by protocol, or by application type. This allows real-time traffic to be given precedence over non-critical traffic.

Adding control protocols and mechanisms: You can add control protocols and mechanisms that help avoid or alleviate the problems inherent in IP networks. Protocols like RTP (Real Time Protocol) and RSVP (Resources Reservation Protocol) can provide greater assurances of controlled QoS within the network. And mechanisms like admission controls and traffic shaping may also be used to avoid overloading a network.

Network warm-up exercises

While the higher-ups in the management are yet to pass your VoIP procurement budget, you can do a few things to prepare your existing network infrastructure for the inclusion of new technology. The priority is to make your LAN as efficient as possible. This can be accomplished in several ways.

The most important way is to get rid of unnecessary protocols. NetWare supports IP now, Windows 2000 no longer requires NetBIOS for file and print traffic, and most mainframes now support IP. Although it may require considerable effort to remove IPX, SNA, NetBIOS over TCP, and other protocols it will result in more available bandwidth on the network, more available memory, and faster response time in the clients and servers.

Once you have removed all these protocols, you may find that the level of broadcasts on your network doesn't justify such small subnets anymore. Moving to larger, 'flat' networks will allow you to remove routers that are no longer needed. These routers are not only points of failure, but are frequent bottlenecks, and usually a major source of delay. Now your VoIP traffic will be more reliable and of high quality.

QoS is a major requirement for VoIP. So why wait until the last minute to QoS-enable your network? Since the configuration of QoS and VoIP are both relatively complex tasks, you don't want to be doing them at the same time if you don't have to. By working the bugs out of your QoS configuration now, your VoIP project will be less likely to encounter glitches.

And last, it's never too early to begin planning. Among the things you can do before you even pick a vendor are:

  • Identify the types of traffic on your network and prioritize them. Voice may actually not be the most important.
  • Determine existing call-traffic statistics and predict future statistics, including cost, average simultaneous calls, average duration, and source/destination pairs.
  • Prepare your network management system for VoIP, including upgrading your RMON probe and protocol analyzers to recognize and decode VoIP.

Packet switching
A traditional voice call through a telecom network opens a dedicated channel of around 64 Kbps in each direction between the two parties. The bandwidth used in the transmission is around 128 Kbps and the switch ports at all the carriers switches are blocked for the entire duration of the call.

Data networks use packet switching by which a connection is open long enough to send a small packet of data from one system to the other. VoIP converts voice into data packets which are dynamically routed on different paths over the network depending on availability and reconverted to voice at the other end.

Standards soup
Like any other technology VoIP is defined by standards. There are various standards out in the VoIP arena like H.323, SIP (Session Initiation Protocol), and MGCP (Media Gateway Control Protocol). Of this H.323 is the most widely used protocol.

Widely-used H.323
H.323 is a suite of protocols developed for specific applications like synchronization, control, and compression codecs. A codec, which stands for coder-decoder, converts an audio signal into a compressed digital form for transmission and back into an uncompressed audio signal for replay.

H.323 allows customer's products to interoperate with other H.323-compliant products and provides standards for interoperability between LANs and other networks. Network managers can restrict the amount of network bandwidth available for conferencing.

A little SIP
SIP (Session Initiation Protocol) is used to establish real-time calls and conferences over IP networks. It is independent of the packet layer, an open standard, and scalable. It has been designed to be a general-purpose protocol. However, extensions to SIP are needed to make the protocol truly functional in terms of interoperability.

Master/slave MGCP
MGCP (Media Gateway Control Protocol) is a master/slave protocol that provides a tight coupling between the MG (Media Gateway) which is the endpoint and the MGC (Media Gateway Controller) server. MGCP-based VoIP solutions separate call control (signaling) intelligence and media handling. In MGCP architecture, the MGC server or 'call agent' is mandatory and manages the calls and conferences and supports the services provided.

Benefit of a converged network
You can create a simple and scalable VoIP architecture with the inclusion of hardware and software elements like VoIP routers, VoIP gateways, switches, and IP phones. In cases where a router or switch does not support VoIP, all one needs to do is install a small chip or just upgrade the software to VoIP-enable these devices. Most vendors offer these upgrades for their equipment.

In case of converged networks, the initial investment may turn out to be slightly higher, but one can observe the ROI on this technology within a very short time span.

VoIPs core benefit is its ability to make next generation converged networks a reality. In a converged network environment, telephony and data signals are transmitted as packets over the data network. A typical office has a separate network for data transmission and voice (telephone).

Now, a converged network enables you to transmit voice over the existing data network. This maximizes the efficiency of your network. The traditional voice circuits can be used as backup or even eliminated.

It also simplifies your network architecture. A single infrastructure is capable of carrying both data and telephony traffic. You don't need to pull separate cables for services. Network deployments and reconfigurations are simplified, and service can be extended to remote sites and home offices over cost-effective IP links.

QoS is the key
The idea of carrying voice traffic over the data network is great. But there are a few issues that need to be considered. Voice is converted to regular data packets and moves in the network just like other data packets. If there are transmission errors the packets need to be re-transmitted, and in case of bottlenecks the traffic is temporarily blocked. The voice packets are thus delayed and conversation is hindered. There are also problems like echo and jitter. Here's where QoS comes in and plays a key role to ensure acceptable call quality.

IP Phones and current generation routers usually have QoS support built into them. A QoS bit is attached to all voice packets as it hits the network. The network devices recognize this extra bit and offer it priority over regular data packets. This way, voice moves first over the network and thus does not cause unnecessary delay. In case of transmission errors which require re-transmission, there is no drop in call quality. Certain audio codecs can compress the header of VoIP packets to reduce its size.

Migration issues
Migrating from an Ethernet LAN gives rise to a few delay issues. Ethernet frames are variable in length, and Ethernet has no mechanism for prioritizing one frame over another. Therefore, as network traffic increases, small frames carrying a voice payload may often have to wait in switch buffer queues behind large frames carrying data. With voice having small delay tolerance, the lack of prioritization across a switched Ethernet network may degrade the quality of voice communications.

The most promising solution is to handle the problem at layer 3 via the RSVP (Resource ReSerVation Protocol). RSVP operates by reserving bandwidth and router/switch buffer space for certain high priority IP packets like those carrying voice traffic. RSVP is still only able to set up paths for high priority traffic on a 'best effort' basis, and thus it cannot guarantee the delay characteristics of the network.

Fast Ethernet and Gigabit Ethernet presents a clearer migration path than ATM. Migrating from an ATM network to a converged VoIP infrastructure may be principally simple because ATM was designed specifically to support both voice and data traffic over a common infrastructure. It also provides multiple QoS levels.

Future VoIP
A future VoIP network will include iPBXs (IP-based PBXs), which will emulate the functions of a traditional PBX. These will allow both standard telephones and multimedia PCs to connect to either the PSTN or the Internet, providing a seamless migration path to VoIP. An iPBX can also combine the features of today's switches and routers and could become the gateway for variety of value-added services like directories, message stores, firewalls and other network-based servers.

Such a VoIP system would also combine real-time and non real-time communications.

Soutiman Das Gupta can be reached at soutimand@networkmagazineindia.com

 
     
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