How to Ensure Smooth Voice-Data Convergence

The way voice is carried over IP networks is fundamentally different from

legacy telephone networks. Traditionally, it has been sent as a continuous

stream over circuit switched networks and the tariff depended on the duration of

the call. But, over IP, voice is broken into packets just like data and sent

across as a continuous stream. The IP network could be your enterprise LAN for

inter-departmental communications or your WAN links for calls to clients or

other branch offices. The voice packets may take different routes through the

network but are assembled back in order at the called end to make up for a

meaningful conversation. Such a convergence of voice and data networks ensures

ease of management for network admins, as they don't have to bother about

maintaining two separate networks. Now, although infrastructure setup for

converging the two networks is a tad costly, it more than makes for that with a

steady drop in call costs over the long run. However, challenges related to

ensuring the reliability and quality of a voice call still need to be overcome

before such convergence becomes a norm in future. The issue is not just about

removing the bottlenecks but more so about ensuring that there is enough

capacity in the network, the quality of calls is above customer expectations and

that there is a balance between service availability and cost. Therefore, you

need continuous voice quality assessments, and planning for new application

bandwidth to ensure a successful convergence.

Packet switching, the fundamental technique in VoIP, allows for a more

efficient use of bandwidth. But it doesn't inherently provide a guaranteed QoS.

Let's understand the process briefly. A router divides a stream of data into

smaller chunks, with the addresses for the originating and destination device in

the packet's header. Based on the packet's destination, the router forwards it

to the next one in sequence (avoiding congested paths). This way they can arrive

at their destination without any significant delay. At the destination, TCP

arranges them in the right order. Now, this approach is good for data such as

e-mail, IMs and peer-to-peer file sharing but not for voice. Any delay in

delivering the words in right order or missing words, render the speech

meaningless. All these problems are further accentuated when voice traffic is

interspersed with data over integrated networks such as Internet.

Voice quality



With voice traveling over the same network backbone as data, it inherits the
same problems as have bugged IP networks in the past. Some of these get even

more pronounced with voice and are absolutely intolerable. VoIP traffic becomes

vulnerable to network delay, jitter, and packet loss, making it a difficult

network technology to manage. Jitter is a sudden variation in the expected

arrival time of a packet and is caused by the network a packet traverses (which

includes transmission medium such as wire, cable, or optical fiber). Another

common concern is latency which is the amount of time delay between the

initiation of a service request for data transmission and the grant of that

request. Delay can cause problems such as echo and speaker overlap. Echoes are

signal reflections of the speaker's voice from the far-end telephone equipment

back to the speaker's ear. These become a major irritation if the delay exceeds

50 ms. Delay induced echo can be overcome by the use of echo cancellation

technology. Still greater problem is that of speaker overlap-one talker stepping

on the other talker's speech. This is caused if delay exceeds 250 ms. Since loop

(round-trip) delays are greater than this value for virtually all VoIP

connections, all VoIP gateways need to have an echo cancellation function.

Delayed or missing packets could be imperceptible for e-mail or other such

apps but absolutely intolerable for quality VoIP communications. As per ITU,

minimum acceptable delay for VoIP calls is 0—150 ms for local calls and 150-400

ms for international calls. Another phenomenon to monitor is packet loss.

Networks either sporadically drop single packets (called gap periods) or large

numbers of packets in a 'burst.' Although packet losses are satisfactorily

managed by packet loss concealment techniques during gap periods, it is the

sustained bursts which are difficult to manage. Managing VoIP quality primarily

involves minimizing network delay and jitter, because codecs require a steady,

consistent stream of packets to provide quality audio.

A jitter buffer on a VoIP phone can mask mild delay and jitter problems but

QoS parameters need to be negotiated up front before the data transfer begins, a

process referred to as signaling. This gives an opportunity to determine if the

required network resources are available and in most cases reserve the required

resources before granting a QoS guarantee to the client.

Bandwidth concerns

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.

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