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Is QoS The Key To Accelerate Deployment Of IP Telephony?

QoS for voice transmission over IP networks is imperative for an experience equivalent to that offered by PSTN

The prophecy of Internet Protocol (IP) becoming the universal carrier for all communication traffic is facing rough weather despite its promising potential. It is not just about picking up the phone and making a call. The concept of a voice carrier across locations having no constrains of geographies whatsoever, is the prediction of the new generation of communication. When the technology behind IP Telephony matured, the immediate challenge was that of the infrastructure. It is now evident is that critical data transmission across this medium will be a larger concern? Migrating onto the IPSTN bandwagon for voice transmission over the existing data infrastructure is a Herculean task in itself.

The mission then for IP telephony is to ensure delivery of the same quality of reliable voice conversations that users have come to expect after a century of experience over the primary public switched telephone network (PSTN) lines. A greater effort is for organizations to view IP telephony as delivering voice service over an IP network such that users have virtually the same experience as when voice services were transported over a circuit-switched network.

As end users demand more quality and reliability, providers have shown more concern towards Internet protocol (IP) testing for end-to-end quality of service (QoS). QoS mechanisms provide a set of tools that can be used by the network administrator to manage network resources in a controlled and efficient manner. These will have the effect of offering improved service to mission critical applications and users, while simultaneously stemming the rate at which capacity must be increased.

In other words, QoS can help improve service to the network users while reducing cost of providing these services. To support IP telephony, a network must guarantee certain QoS metrics, such as an end-to-end delay of roughly 150ms or less and a delay variance (jitter) of no more than 30ms. To ensure that these metrics are met, network operators must invoke the most reliable QoS mechanisms, which will typically include ways to prioritize and queue traffic or reserve network resources end-to-end.

Understanding IP Telephony
The term "IP telephony" covers a range of technologies, including Voice over IP (VoIP) and Fax over IP services, which are carried over both the Internet and private IP-based networks. IP telephony is part of packet voice that includes Voice over Asynchronous Transmission Mode (VoATM) and Frame Relay networks, which run faster than IP but are less common. IP telephony connects across combinations of PCs, Web-based telephones and phones connected via public telephone lines to remote voice gateways. Because information travels in discrete packets, it need not rely on a continuously available switched circuit. Consequently, it's easier on bandwidth and is also cost efficient.

IP telephony is still a "best effort" system challenging the "guaranteed" quality of service provided by the Public Switched Telephone Network (PSTN). While the Internet best-effort service model continues to treat all network traffic in exactly the same way, there is no consistent service outcome from the same. When the load level is low, the network delivers a high-quality service. The best-effort Internet does not deny entry to traffic, so, as the load levels increase, the network congestion levels increase, and service quality levels decline uniformly. This decline in service is experienced by all traffic passing through a congestion point, and is not limited to the most recently admitted traffic flows.

QoS is the key to accelerating the deployment of IP telephony. Although IP telephony is an appealing technology, QoS had been concern enough to limit adoption of this technology. The two most important components of QoS, latency and lost packets, relate closely to each other.

Every VoIP application involves converting speech into series of packets, each containing about 30 of voice. After the speaker-side gateway receives the voice transmission, it converts it to packets and shuttles these packets onto the network. Packets traverse the network and reunite at the receiver's end. The network may lose some packets, and others arrive too late to use in the reconstructed speech. In either case, the speech plays back without these packets lost in transit.

Processing and transmission delay all packets, and this delay causes latency in conversations. The transmission leg across the network is the longest, especially on the Internet. Users expect latency of 250ms or less, equivalent to the delay of a satellite link for international calls for "toll-quality" service. Unfortunately, the Internet induces latencies that can far exceed 500ms.

Jitter buffers also contribute to latency. Jitter, or delay variation, is the result of packets arriving at their destination at irregular intervals. Bursts of Internet traffic create jitter problems. This distortion is particularly damaging to the QoS of VoIP applications. Severe jitter in IP voice transmissions causes jittery or shaky voice quality. We can also understand jitters as the speed variation between quickly and slowly traveling packets. The jitter buffer stores packets, allowing most of the slower packets to catch up. The less control in routing, the more jitter that results, and more jitter means a longer jitter buffer. But a longer jitter buffer introduces more latency. Too short a jitter buffer loses too many packets, causing voice quality to tumble.

When the network loses a packet, VoIP products "reconstruct" it. The products cannot determine the information in the packet, but like CD players smoothing over scratches, VoIP algorithms produce transitions that are less distracting than silence. Yet, too many lost packets degrade voice quality to unacceptable levels. The maximum level of lost packets for toll-quality service is difficult to define, but 10% is common.

Maximizing Throughput
Latency, loss of packets and distortion are critical 'cannot haves' in an IP network. The industry has evolved from building interfaces between legacy systems and 'New World' technologies to one with solutions and products that directly address communication over a leased line and between closed user groups. These solutions would translate to routers, soft switches and IP enabled equipment populating the network between the source of data transmission and the destination. What is essential is a maximum throughput despite transportation losses especially over large networks like the Internet itself. The throughput rate decreases with increase of delay and packet loss.

QoS: A Mission Critical Criteria
The objective of various Internet QoS efforts is to enhance this service with a number of selectable service responses. These service responses may be different from the best-effort service by some form of superior service response, such as lower delay, lower jitter, or greater bandwidth. QoS service responses may be distinguished by providing a consistent, and therefore predictable, service response that is unaffected by network congestion levels.

These are quantitative service responses, where the characteristics of the service can be measured against a constant outcome. A quantitative service many be one that constrains jitter to a maximum level, or one that makes a certain bandwidth available, within parameters of bounded jitter, similar to a conventional leased line. Such constant-rate services may be superior to best-effort services when the network is under load, but they may also offer inferior service when the network is under negligible load. The essential attribute of these services is one of consistency.

Why is there a need for relative or consistent service profiles within the Internet? First is the desire to provide high-quality support for IP voice and video services, second, the desire to manage the service response provided to low-speed access devices, such as Internet mobile wireless devices, and third is the desire to provide a differentiated Internet access service, providing a network client with a range of service-quality levels at a range of prices.

The initial approach to QoS was that of the Integrated Services architecture where the request is sent to the network as a reservation. The network in turn would respond with its ability to carry the additional load by committing to the reservation. Termination would mean another request or a signal from the network corresponding its inability to continue the reservation. The essential feature of this model is the "all-or-nothing" nature of the service model. This approach imposes per-application state within the network, and for large-scale networks, such as the global Internet itself, this approach alone does not appear to be viable.

The subsequent approach was to examine those mechanisms that can provide differentiated service outcomes with appropriate scaling properties. The differentiated services architecture includes dropping the concept of a per application path state across the network using instead the concept of aggregated service mechanisms. Within the aggregated service model, the network provides a smaller number of different service classes and aggregates similar service demands from a set of applications into a single service class.

Aggregated services are typically seen as an entry filter, where on entry to the network, each packet is classified into a particular service profile. This classification is carried within the IP packet header, using six bits from the deprecated IP Type of Service (ToS) header to carry the service coding. The network then uses this service code in the packet header to treat this packet identically to all other packets within the same service code.

Neither of the two approaches has proved adequate. With one approach imposing an excessive load in the core of large networks through the imposition of a per-application path state and the other while providing superior scaling properties through the use of aggregated service elements, includes no concept of control signaling to inform the traffic conditioning elements of the current state of the network, or the current per-application requirements, the question is whether a combination of the two approaches would be the answer.

The telecommunications industry has been challenged. Voice moved from analog to digital translation and transportation over a PSTN, making the geographical divided insignificant. Telecommunication has traversed extensively. However, challenges defy innovations—be they technology or otherwise. Global players are running this race to make voice a run of the mill entity in the telecom space. Voice will ride the tube with data and will be just another application on the global networks.

Will tech gurus then speculate 'free voice for all' as a future? This is one question technology and the burgeoning riders of the new Telecom Arc will have to address at haste.

As end users demand more quality and reliability, providers are showing more concern towards IP testing for end-to-end QoS

QoS mechanisms provide a set of tools that can be used by the network administrator to manage network resources in a controlled and efficient manner The mission then for IP Telephony is to ensure delivery of the same quality of reliable voice conversations that users have come to expect after a century of experience over the PSTN lines

Naresh Wadhwa, Head-SP Sales, Cisco India, can be reached at nwadjwa@cisco.com

 

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