WildPackets VP of marketing Tony Barbagallo swears that VoIP expletives - jitter and packet loss as well as latency - can be successfully deleted by following his advice below:
 Traditional telephone services have typically gained a reputation of providing excellent voice quality and superior reliability. Consequently, users take for granted that their phone systems will provide high quality with virtually no downtime. Yet many VoIP installations fail to meet these expectations primarily because organizations have not adequately evaluated their network infrastructure to determine whether it can adequately support applications that are very sensitive to: |
The VoIP expletives
1. Jitter
2. Packet Loss
3. Latency
VoIP, as well as other media applications, require a steady, predictable packet delivery rate in order to maintain quality.
Jitter, which is variation in packet delivery timing, is the most common culprit that reduces call quality in VoIP systems. Jitter causes the audio stream to become broken, uneven, or irregular. As a result, the listener’s experience becomes unpleasant or intolerable.
The end results of packet loss are similar to those of jitter, but are typically more severe when the rate of packet loss is high.
Excessive latency can result in unnatural conversation flow where there is a delay between words that one speaks versus words that one hears. Latency can cause callers to talk over one another, and can also result in echoes on the line.
So, the impact of jitter, packet loss, and latency can be severe.
Some VoIP systems are all but unusable from the time they are implemented, because no one took the time to properly profile the existing network. Before implementing VoIP, a thorough application impact study should be completed.
The following five steps are critical to deleting the VoIP expletives:
| Step 1. | Establish a thorough baseline of current network activity on all segments that will host VoIP. Take a close look at access point loads as network activity can introduce inconsistency in packet delivery rates and cause VoIP late packet arrivals. |
| Step 2. | Analyze store-and-forward and queuing congestion in switches and routers which can lead to packet spacing unpredictability and thus jitter. Keep in mind that the more hops a packet has to travel, the worse the jitter; reduce hops as much as possible. |
| Step 3. | Accurately estimate the network resource utilization for the proposed VoIP system. Unlike traditional land lines where users typically have dedicated switch ports, a VoIP system is a shared medium. Be sure to consider the number of simultaneous calls that will occur. The maximum number of VoIP calls per 802.11b access point (the majority of currently available VoWLAN handsets) ranges from 5 to 13 depending on Codec, header type and operating rate. |
| Step 4. | Select a Codec with higher speech sampling rates. In general, the higher the speech sampling rate the better, but the more bandwidth consumed. Make tradeoffs carefully. |
| Step 5. | Verify that Quality of Service (QoS) is supported on all segments and devices over which VoIP traffic will travel. |
Only when all of these topics are addressed can the feasibility of VoIP be assessed. The outcome of these analyses may indicate that VoIP can be supported. However, the results may point to aspects of the network that need to be modified before they will be truly ready for VoIP. While all of these steps seem like a lot of work, they are the only means to assure that your VoIP system will not be crippled by adverse network characteristics.
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View VoIP solutions from WildPackets
Have you gone thru the 5 steps?
If so, what tips/lessons learned can you share?
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