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Accelerator products can help enterprises a lot in addressing the performance requirements of all enterprise applications including voip. First, Accelerators change the economics of wide area networking by squeezing an average of 100% – 400% more bandwidth with peaks of 1000% depending on traffic mix. This frees up link bandwidth to support high quality voip services – and it does it without expensive WAN upgrades.

It is also worthy to note that Accelerators do not actually use lossy compression schemes that might degrade voice quality and they add less than one millisecond of latency. In fact, the Accelerator’s compression actually reduces end-to-end latency by reducing serialization delays on WAN links. For example, it takes 125 ms to serialize a 1,000 byte packet on a 64 kbps link, but if an Accelerator increases the effective bandwidth by 4X to 256 kbps, the serialization delay is reduced by a factor of four to 31 ms. The following formula can be used to calculate the serialization delay for any combination of packet size and link speed:

Packet Size (in bytes) x 8 / Speed (in kbps) = Serialization Delay (in ms)

In addition to freeing up the bandwidth normally consumed by data applications, Accelerators are able to reduce WAN bandwidth requirements for different voip codecs. In fact, tests have shown that Accelerators reduced G.711 bandwidth requirements by 20% and G.729 by 70%. As a result, WAN links can carry more simultaneous voice calls and the performance of other applications may also be improved.

Accelerators solve increased jitter and latency caused by large data packets over slow WAN links by fragmenting large data packets and injecting voip packets at regular intervals. This feature allows voip and data to co-exist even on branch office WAN links. For example, normally, a voice over IP packet “stuck” behind a 1,500 byte packet on a 64kbps lin will be delayed by 188ms.

Using the Accelerator’s packet fragmentation will result in the data packet being reduced in size (accelerated – say from 1500 bytes to 500b bytes) and then fragmented into smaller data packets (say – 2 packets of 250 bytes each). In this case, the latency for the voip packet will go down from 188 ms to 31ms! In addition to increasing WAN capacity for both data and voip while reducing latency and jitter, Accelerators also manage WAN bandwidth to ensure that critical applications like voip get the bandwidth they need.

Accelerators include an Instant QoS feature that prioritizes application access to WAN bandwidth. Without such priorization, the additional effective bandwidth provided by Accelerators could be consumed by aggressive, non-critical applications such as file sharing. Accelerator’s AppView feature provides graphical visibility for all application traffic sharing a link. AppView can be used to monitor WAN utilization and to plan future capacity requirements.

And finally, ip accelerators have a set of data integrity features that are designed to stop the packet loss that can degrade voice quality. A flow control mechanism reduces packet loss caused by link congestion and a packet recovery feature ensures that any lost packets are transparently recovered at the link level before they can cause voice quality problems.

Jim Francisto

You can determine how much voip bandwidth to set aside for voice traffic using simple math. However, in a converged voice and data network, you have to make decisions on how much voip bandwidth to give each service. These decisions are based on careful consideration of your priorities and the available voip bandwidth you can afford. If you allocate too little voip bandwidth for voice service, there might be unacceptable quality issues.
Another consideration is that voice services are less tolerant to voip bandwidth depletion than that of Internet traffic. Therefore, voip bandwidth for voice services and associated signaling must take a priority over that of best-effort Internet traffic.
If a network were to use the same prevailing encoding (CODEC) scheme as the current PSTN system, voip bandwidth requirements for Voip networks would tend to be larger than that of a circuit-switched voice network of similar capacity. The reason is the overhead in the protocols used to deliver the voice service.
Typically, you would need speeds of OC-12c/STM-4 and higher to support thousands of call sessions. However, Voip networks that employ compression and silence suppression could actually use less voip bandwidth than a similar circuit-switched network. The reason is because of the greater granularity in voip bandwidth usage that a packet-based network has in comparison to a fixed, channel size TDM network.

Allocations of network voip bandwidth are based on projected numbers of calls at peak hours. Any over-subscription of voice voip bandwidth can cause a reduction in voice quality. Also, you must set aside adequate voip bandwidth for signaling to ensure that calls are complete and to reduce service interruptions. The formula for calculating total voip bandwidth needed for voice traffic is relatively straightforward. The formula to calculate RTP bearer voice voip bandwidth usage for a given number of phone calls is as follows:

Bits per sec = packet creation rates per sec x packet size x number of calls x 8 bits per sec

Where samples per sec = 1,000 ms / packet creation rate

Example: 2,000 full-duplex G.711 encoded voice channels that have a packet creation

Rate of 20 ms, with a packet size of 200 bytes (40 byte IP header + 160 byte payload)

50 samples per second = 1,000 ms / 20 ms

160 Mbps = 50 x 200 x 2,000 x 8

Note that this number is a raw measure of Voice over IP traffic and does not take in account the overhead used by the transporting media (links between the routers) and data-link layer protocols. Add this raw IP value to that of the overhead to determine the link speeds needed to support this number of calls. Note this value represents only the bearer (voice) content. Signaling voip bandwidth requirements vary depending on the rate at which the calls are generated and signaling protocol used. If a large number of calls are initiated in a relatively short period, the peak voip bandwidth needs for the signaling could be quite high. A general guideline for the maximum voip bandwidth requirement that an IP signaling protocol needs is roughly three percent of all bearer traffic. Using the previous example, signaling voip bandwidth requirements if all 2,000 calls were initiated in one second would be approximately 4.8 Mbps (3 percent of 160-megabits).

With the calculation of bearer and signaling, the total voip bandwidth needed to support two thousand G.711 encoded calls would be an approximate maximum of 164.8 MB. This ip bandwidth requirement is a theoretical maximum for this specific case. If the parameters change, such as call initiation rate, voice encoding method, packet creation rate, employment of compression, and silence suppression, the voip bandwidth requirements would change as well. With large Voip implementations requiring sizable voip bandwidth, it becomes imperative that the IP network delivers the needed service at predictably high performance.

Jim Francisto
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