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Feature Article
May 2004

Is Your Network Ready For Convergence?


Is your data network ready to carry packetized voice traffic on top of all your current applications? For most enterprise network managers, the likeliest answer would be: I don�t know. That�s the make-or-break question for many network executives as they consider the promise of integrated voice/data.

The biggest challenge in transitioning from traditional circuit-switched voice and video systems to the new, more economical voice and video over IP packet-switched technologies is obtaining adequate quality of service (QoS) over the network. Quality of service is the capability built into the network to guarantee that information traverses the network in a timely manner. Most existing data networks were designed for bursty applications that are delay-insensitive, meaning that if a data packet arrives within a reasonable amount of time, both the application and the user are satisfied.

Voice and video data, on the other hand, are very sensitive to delay; if a packet arrives more than approximately 170 milliseconds (ms) after it is transmitted, the packet is worthless as it will arrive too late to be used in the conversation or video image. Consequently, networks carrying IP voice must be designed and configured properly to ensure that real-time packets traverse the network efficiently.

The challenge of obtaining adequate quality of service is exacerbated when a data packet must traverse the WAN. Typical local-area networks (LANs) run at 10 Mbps, 100 Mbps, 1000 Mbps, and higher. However, because bandwidth over the WAN is significantly more expensive than over the LAN, many wide-area networks operate at T1 speeds (1.45 Mbps) and slower, creating a huge bottleneck at the LAN/WAN interface.

For normal data packets like e-mail, Web browsing, client/server programs, and a host of other applications, this LAN/WAN bottleneck is a nuisance, but not an application killer. However, when voice and video packets must compete with regular data packets for transmission over a bandwidth-constrained WAN, voice and video applications are often rendered useless.

For IP networks supporting voice, video, and data applications, the network quality of service is evaluated by measuring four key parameters: bandwidth, end-to-end delay, jitter, and packet loss.

  • Bandwidth: The average number of bits per second that can travel successfully through the network.
  • End-to-end delay: The average time it takes for a packet to traverse the network from a sending device to a receiving device.
  • Jitter: The variation in end-to-end delay of sequentially transmitted packets.
  • Packet loss: The percent of transmitted packets that never reach the intended destination.

Target values for delay, jitter, and packet loss are <170 ms, <50 ms, and <1% respectively. Organizations wishing to maintain management control of their networks when adding new applications, including voice and video over IP, usually implement control procedures using a number of available technologies.

There are a number of control-based measures to provide adequate quality of service.

IP Precedence and DiffServ
Both of these traffic-marking schemes modify and mark certain bits in the data packet header. Upon arrival at an IP precedence or DiffServ-enabled router or switch, packets with the header bits set appropriately are given priority queuing and transmission.
In a network controlled by packet marking, voice packets would be given the highest priority since they are very sensitive to delay and jitter, even though voice is not particularly bandwidth-intensive.

Queuing occurs in routers and switches. Different queues or buffers are established for the different packet-marking schemes. One of the queues, for example, might be established for delay- and drop sensitive information like voice and video data. Voice and video packets marked with certain IP precedence or DiffServ values will be placed in these high-priority queues.

Queuing-based solutions have a number of drawbacks. Of these, one of the most significant is the lack of any feedback mechanism for determining how applications are competing for bandwidth. Consequently, data traffic for applications on networks with queuing mechanisms in place cyclically ramp up and back off transmission rates based upon packets being discarded. This causes chunks of data that accumulate at the LAN/WAN interface where speed conversion occurs.
One way to eliminate these chunks of data is by using a special technology called TCP Rate Control. TCP Rate Control paces or smoothes network data flows by detecting a remote user�s access speed, factoring in network latency, and correlating this data with other rate and priority policies applied to various applications. Rather than queuing data in a switch or router and metering it out at the appropriate rate, TCP Rate Control induces the sending applications to slow down or speed up, thus sending data just-in-time. By shaping application traffic into optimally sized and timed packets, TCP Rate Control can improve network efficiency, increase throughput, and deliver more consistent, predictable, and prompt response times.

Most voice and video applications use UDP rather than TCP for transmitting real-time communications data. Unlike TCP, UDP sends data to a recipient without establishing a connection, and UDP does not attempt to verify that the data arrived intact. Therefore, UDP is referred to as an unreliable, connectionless protocol. The services that UDP provides are minimal � port number multiplexing and an optional checksum error-checking process � so UDP requires less processing time, and lower bandwidth overhead than TCP. This allows UDP packets to traverse the network more rapidly, which is a desirable characteristic for voice and video applications.

However, because UDP doesn�t manage the end-to-end connection, it does not get feedback regarding transmission conditions; consequently, applications transmitting UDP packets cannot prevent or adapt to congestion. Therefore, UDP can end up contributing significantly to an overabundance of traffic impacting all traffic on the network. This may cause latency-sensitive flows, such as voice and video over IP, to be so delayed as to be useless. In these instances the voice or video application may still continue to transmit data, oblivious to the fact it is contributing to the delay problem.

A process called �partitioning� can control the rate of UDP transmissions over the WAN.

Partitioning is a special case of rate control in which specific amounts of bandwidth are set aside for the most important classes of traffic. Partitioning can also be overlaid on top of TCP rate control for TCP based applications. A packet-shaping appliance that examines all packets traversing the network administers partitioning. By identifying a particular application as a member of a particular partition class, the Shaping Appliance is able to control how much bandwidth each application or class of applications uses, and it can ensure that a particular partition always gets sufficient bandwidth.

When the bandwidth within a particular partition is not fully utilized, the excess bandwidth can be reallocated to partitions serving other important applications. Administrators can also specify UDP flow maximum mechanisms so that one large flow (e.g., video conferencing or streaming video) does not consume all bandwidth on the network. In environments where multiple video devices are deployed, organizations will want to couple partitioning with a SIP proxy�s or H.323 gatekeeper�s call bandwidth controls to manage how many simultaneous video calls can be placed.

Transitioning to voice and video over IP requires rock-solid quality of service over the wide-area network link. One of the major IP packet congestion points is at the LAN/WAN interface, where bandwidth availability can often decrease by two or more orders of magnitude. Provisioning additional WAN bandwidth may provide temporary relief, but it may be an expensive short-term solution because additional bandwidth is often consumed by more aggressive non-business applications like Web surfing and peer-to-peer file sharing.

Arvind Ahuja is senior product manager for Packetwise software at Packeteer, Inc., a leading provider of application traffic management solutions designed to enable businesses to gain visibility and control of networked applications, extend network resources and align application performance with business priorities. For more information, visit the company online at www.packeteer.com.

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