How to Ensure Smooth Voice-Data Convergence

A converged network is easy to manage and reduces costs over a period of time. But spare a thought for end users who might sulk over jittery/delayed conversations just because their call got lost somewhere along the networking maze. How to ensure smoother operations? Let's look at the core issues and the technologies available to resolve those

Saturday, March 10, 2007

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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.

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