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How to Ensure Smooth Voice-Data Convergence

Continued from page: 1

Saturday, March 10, 2007

Bandwidth concerns
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Bandwidth concerns
VoIP has the potential of providing CD quality audio to its users. But to achieve that, you would be putting enormous strain on the available bandwidth. Most of the users would be more than happy to settle for a low bandwidth, glitch-free call quality. Therefore, simply figure out how much additional bandwidth is required and how much of your network needs an upgrade to facilitate a meaningful conversation. With enterprises upgrading their WAN infrastructure in a big way, that should not be much of an issue. Looking a bit further, you can see technologies that unify the various modes of communication such as messaging, video and collaboration, being integrated over your IP networks. So you can foresee that not only does your bandwidth needs to be increased, but utilized properly as well. The last thing you want is a lot of angry users complaining about call quality just because your bandwidth is either overloaded or not utilized properly.

Other causes for delays could be pathway congestion, time taken for error checking, transmission negotiations and additional info to determine the type of data being sent, its origin and destination. What this means is that enough bandwidth must be made available to allow for not only voice transmission but also the extra bandwidth for overheads required for any data transmission. The actual amount of bandwidth for voice also depends on the codec used for compression. This can range anywhere from 16 – 64 kbps and after compensating for overheads, safely assume a total of 88 kbps.

Mean Opinion Score (MOS)
This is a traditional way of estimating call quality. MOS measurement is determined by a group of listeners who rate the quality of audio based on samples played. The MOS rating system has a five-point scale ranging from 1 to 5 where 1 means poor quality and 5 translates to excellent. Practically, even if you achieve a score of 4, you would have done your job nicely. However, anything below 3.5 should set alarm bells ringing. But there's a catch. Call quality may be perceived differently depending on the environment. For example, a user making a call from some noisy environment would tolerate a far lower quality than an executive taking a call in a conference room. Using this technique, problems can be addressed when voice quality starts to degrade before the users feel any debilitating effects. MOS is also useful in troubleshooting and in capacity planning as you anticipate increases in call volume over time.

Traffic control
Traffic shaping is a technique to control the network traffic to ensure low latency, enforce policies and ensure optimum utilization of bandwidth. This is done by controlling the volume and the rate at which data is sent through a transmission path. Most of the traffic shaping schemes are implemented at the network edges to control traffic entering the network. You can control the network dynamically, prioritizing bandwidth amongst applications, depending on the rise and fall in network usage. There are several other ways in which WAN bandwidth can be dynamically managed:

  1. Set rules as per applications: Traffic shaping solutions can categorize traffic in terms of priority, thereby assigning a specific amount of bandwidth for each. You might just assign a rule that limits aggregate FTP traffic to no more than 6 Mbps and another one that limits total streaming audio traffic to no more than 3 Mbps. This categorization can also be on the basis of the traffic's protocol and the ports used by an application. You can also categorize traffic based on the content. Most traffic shapers categorize web traffic based on the interactions between a web server and a browser when a page is requested, regardless of the port number.
  2. Set rules for each user: Traffic shapers can set traffic limits for each user so that traffic is shared fairly amongst all users. For instance, you might decide to limit traffic to or from each user to no more than 256 Kbps. This way although a user can access whatever he wants, but the traffic flow is smoothed out to a specified level rather than hogging the total available network capacity unnecessarily. Traffic limits can be set to be either hard or burstable. A hard limit is always fixed and can't be avoided at any cost. Burstable limits allow traffic to exceed a threshold value (called 'burst limit'), if you have spare capacity with no higher priority application to load the vacant capacity with.
  3.  Traffic priority management: Other than setting hard or burstable traffic limits on applications or users, traffic shaping devices can also define the importance or priority of different types of traffic. In a converged network, voice packets can be given a higher priority over regular data such as e-mail, IMs, peer-to-peer file sharing and so on. Some traffic shaping tasks can be done directly on a router, just as you do firewall-like packet filtering. However, using specialized traffic shapers avoids loading up routers, leaving them free to focus on routing packets as fast as they can.

By carrying out simple tasks such as managing the number of calls placed across an IP WAN link, network managers can ensure that their network's voice/data quality remains high. Using call admission control they can limit the voice bandwidth and if necessary, allow calls to be routed across the PSTN to ensure good quality. Also by implementing change control, they can prevent users from using a high-bandwidth application which could harm voice traffic. And of course, through the use of virtual LANs voice traffic can be separated from other broadcast-intensive apps.

A LAN setup based on 802.11pq. Here the systems at the center constitute a data only network while the IP Phone and the Softphone form a converged network

Other techniques
There are several other technologies that can be deployed to enhance or add QoS features to converged networks. Here are a few of them:

  1. MPLS: Multi-Protocol Label Switching complements IP technology by taking advantage of the intelligence associated with IP routing techniques. It specifies mechanisms to manage traffic flows amongst different hardware, machines, or even applications. Data transmission occurs on label-switched paths (LSPs)-a sequence of labels at each and every node along the path from the source to the destination. A label contains information such as destination, precedence, VPN membership, QoS information from RSVP and the route to be followed. These LSPs are established either prior to data transmission or when a certain flow of data is detected. High-speed switching of data is possible because fixed-length labels are inserted at the beginning of a packet and can be used by hardware to switch packets quickly between links. MPLS is rapidly emerging as a core technology for next-generation networks as it allows for the consolidation of multiple disparate networks into one and provides a means for multiple Layer 2 technologies to be used simultaneously, thereby saving costs.
  2. DiffServ: Differentiated Services is a networking architecture which allows you to prioritize packets in the network path through relevant information in their headers. Each data packet is divided into a given number of traffic classes, rather than differentiating traffic based on individual flow. Each router on the transmission path is configured to prioritize traffic based on its class. The DiffServ technique does not automatically prioritize traffic, but follows what has been defined by the network administrator. It also recommends a standard set of traffic classes to ensure interoperability amongst hardware and networks. Since DiffServ is a mechanism that allows you to decide what packets to carry and what to drop; depending on the network capacity, you almost invariably land into a situation where you are on the brink of your WAN link. And since Internet traffic is bursty, this might result in your low priority data being dropped almost always. To avoid this situation, fix a cap on the amount of bandwidth for higher priority data.
  3. 802.11pq: This IEEE protocol describes how switches can classify frames at the Ethernet layer. It also allows end points and routers to assign priorities to LAN frames. It is deployed in small LAN setups to achieve good quality of service-be it voice, peer-to-peer file transfer or IM. If you use a switch or router with 802.11pq capability, you don't have to worry about creating different virtual LANs for your converged network. Such devices are capable of differentiating voice and data frames on their own and can create a subnet. So, let's say you have a network in which ten machines talk data and another 10 which talk both voice and data, then the switch will automatically create two VLANs-one for Voice with ten ports and another with twenty ports for data.
  4. RSVP: Resource Reservation Protocol (RSVP) is a network control protocol that enables Internet apps to obtain differing QoS for data packets. It recognizes the fact that there are certain apps like voice which require both timeliness of delivery and quality while others such as e-mail or file sharing that require reliability of delivery and not necessarily timeliness. It is not a routing protocol but works in conjunction with other protocols. It is used by routers to communicate QoS requests to all nodes along the path of transmission. RSVP capable routers communicate with policy servers within the network, to determine what apps would be granted network resources and which requests will be prioritized, in case there are insufficient resources to satiate all.

By keeping a close watch on the network parameters that affect VoIP, you can take care of infrastructure problems before they lead to poor quality or downtime. Understanding what to monitor and having a thorough VoIP analysis, makes this crucial task much more manageable.

Adeesh Sharma

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