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Roaming occurs when a handset moves out of the range of one access point into the range of another. It gives users the mobility to move around within a local coverage area and still be connected to the network. However, roaming is one of the primary reasons why users experience problems on wireless networks. Excessive roaming times lead to poor quality for voice and video over wireless and can lead to dropped calls or data connections.

Roaming usually involves a channel change, but that depends on the type of technology deployed. If it's a multi-channel architecture, which is most likely the case, a channel change is required. When roaming occurs, the client needs to be re-authenticated and re-associated with the new access point, which takes longer than 150 milliseconds in most instances, especially when advanced features like WPA2 and WMM are in use. Most organization's wireless networks are outfitted with multiple access points (APs) and users can experience poor signal strength and performance despite proper coverage in the area if the client is connected to the "wrong" AP. Even in the most modern, centrally managed systems, the wireless client is the one who decides when to switch from one AP to another. This decision is typically determined based on the current signal strength and is executed by the underlying software controlling the wireless client radio (the "supplicant"). This software is different from manufacturer to manufacturer and from device to device, so the way the decision to roam is made varies widely. In most cases, the wireless client will wait too long and as a result the available signal strength lowers, before the client switches to an AP with greater signal strength.

New and improved standards are available that specify the conditions for "fast roaming," enabling transitions that take as little as 5 - 10 milliseconds. These specifications include:

• 802.11i - with opportunistic key caching so there is no re-authentication step
• 802.11r - fast BSS transition, which optimizes the hand-off as clients move from one access point to another
• 802.11k - radio resource management of WLANs allows re- authentication to be maintained between multiple APs and has predictive capabilities

These new standards (802.11i isn't new, but it's still part of an improving situation for roaming) allow APs greater control in determining when roaming should occur and the APs are more in tune with the current performance of, and demands on, the wireless network. However, this situation is even better when the overall wireless network is under the control of a centralized manager. The issue is that adoption of 11k and 11r has been very slow, especially in wireless clients, and until adoption increases significantly users will continue to suffer slow AP transitions when roaming, leading to poor voice and video over IP performance.

In the meantime, the best approach is to carefully monitor and analyze the roaming activity on your network. Obtaining a complete and accurate view requires real-time aggregation of data from multiple channels and APs, with integrated analysis that leads to detailed reporting - who is roaming, how long each event is taking and what does the average look like for each AP. The end result is simple, yet the process is complex, demonstrating why proper network analysis tools are key to staying productive. 

Telepresence is a big hit in the business world. For good reason too, not only does it provide unique and sophisticated remote collaboration but it's inexpensive and more environmentally friendly. Enterprises are quickly discovering the business value in telepresence as it transforms conferencing into a lifelike and immersive experience with high-definition video, life-sized video displays, high quality sound and unobtrusive cameras and microphones. Cisco, who owns the mindshare around telepresence, couldn't be more pleased.

With the technological benefits of telepresence comes the responsibility in making sure networks can handle the demand. Monitoring the network is critical as it carries the service for telepresence.

In terms of network strain, video is much more demanding than VoIP. VoIP expects regular delivery but video does not as it largely depends on movement and what is going on in the environment. Video is much more bandwidth intensive as well as more sensitive to latency, jitter and packet loss. In fact, below a certain quality threshold for video it is nearly impossible to see anything.

Organizations should understand their current baselines and network in preparing for telepresence. Key metrics include travel levels per segment, packets per second and packet size distribution. Also be aware of what the traffic level is per application including average rates, peak rates and weekly patterns. With this depth of understanding of network performance, organizations can better scope telepresence computing needs and validate network configurations.

There are three points to consider when implementing telepresence in a network environment:  

1) Success breeds success

In order for employees to accept a new technology and adopt it fully, the initial experience using it has to be successful. Beat the outset; be diligent in terms on network analysis as this is when user opinions are formed. Expect the demands to be extreme when first introducing telepresence into the organization.

2) Success increases network demands

The more people that are using the successful technology, the more demand on the network. It's important to stay in tune and not assume a certain baseline status will be reached. Network needs will constantly change depending on load.

3) Interoperability is key

Ensure quality across networks segments you don't control. When monitoring and optimizing the quality of your network, try to reach for the best quality at the end point and don't settle for the lowest common denominator.

Telepresence can provide an organization a lot of advantages and benefits provided the network is designed to handle it. In short, don't compromise on what's under the hood - the network and tools to keep it humming. Otherwise, your telepresence deployment will be remembered as an expensive mistake.

 

Online mobile VoIP (or VoFi) is coming. In-Stat anticipates 171.3 million users by 2013, with annual revenues projected at $10.8 billion ("Mobile VoIP - Transforming the Future of Wireless Voice; In-Stat In-Depth Analysis," Frank Dickson, Sept. 2009). Previously on our blog we've talked about why VoFi and why now, specifically the benefits of VoFi. Now we'll focus on VoFi monitoring, analysis, and troubleshooting.

Before you panic, take a deep breath. Analyzing VoFi traffic is basically the same as analyzing VoIP traffic. Remember though that wireless exacerbates factors such as jitter, latency, and packet loss that affect VoIP. Watch Using VoIP Metrics to Identify Network Problems for the specifics.

Begin at the Beginning: Your End User's Call

When problems arise with VoIP or VoFi applications, you start in the same place. Your first step - before you begin to worry about statistics or packets - is to take the time to listen to representative calls. You want to hear what your end users are experiencing. Your ear will reveal telltale signs of latency, jitter, and packet loss. Be sure your VoIP analysis application supports playback of call audio, specifically the playback of individual RTP streams as well as the playback of the complete call. Without the audio, you can spend hours tracking down problems that aren't due to either the application or the network - for example, clicking due to a damaged handset.

Take Your Network's Pulse

Once you have listened to the call, you'll want to take a look at what's going on in your network.

Figure 1: Overview of Network Health

Immediately you see what you heard - the call quality was poor. The Mean Opinion Score graph gives an average over all calls occurring on your network. In this example there's just one call, so you see the average for the duration of that call.

Dig Deeper

With Expert Events you're able to verify what your ear told you.

Figure 2: Event Summary

With this call, you can see that there are a lot of physical errors: late packet arrival, retries, out of sequence packets, packet loss, excessive jitter, and more. With the cause identified, you can quickly begin to fix the problem. Looking at the call in its entirety, you'll notice the call is closed, it had a successful ending - meaning the call wasn't truncated - what CODEC was used, how long it was, and what the Mean Opinion Score was.

Figure 3: Call Statistics

In this example, the mean opinion score of 2.5 lets you know that the quality of the call was pretty poor. In the media view, you can drill down into each segment leg to determine why the quality was poor.

Figure 4: Call Details - R Factor, Mean Opinion Score, Packet Loss Percentage, One Way Delay, Etc.

Understand the Differences between Wired VoIP and VoFi Calls

The next two figures show both a Wired VoIP call and a VoFi call packet-by-packet. (For an in-depth discussion of these calls, watch Anatomy of a VoFi Call: Packet-by-Packet.) You'll notice that they're pretty similar. The protocols used are different and with VoFi there's the additional step of authentication.

Figure 5: The Anatomy of a Wired VoIP Call

The differences involve: wireless segments instead of wired segments; signal interference; and wireless roaming.

Figure 6: The Anatomy of a Mobile VoIP (VoFi) Call

Learn More

Last week in Toronto, Joe Habib, Director of Global Services, presented "QoS of IP Telephony: Slaying the Three-Headed Beast of Jitter, Latency, and Packet Loss" at IT360. His presentation (PDF) is now available online. If you're interested in ensuring QoS for your current (or future) VoFi deployment, you should definitely check it out.

In the presentation, you will learn:

• What six factors contribute to poor voice quality
•  How to establish metrics for evaluating VoIP call quality
• How to balance high-speed, bursty data requirements with requirements of high quality voice calls
• How to capture data for VoFi Analysis and use VoIP metrics to identify developing problems
• How to analyze a VoFi call packet-by-packet and verify voice quality with call playback

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 met 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 latency, packet loss, and jitter. These three "monsters" that may remain unnoticed in a data network will become very obvious and ugly in VoIP systems.

If you're interested in improving your current (or future) IP telephony experience, check out Jay Botelho's (Director of Product Management at Wildpackets) slide presentation that focuses on best practices to troubleshooting converged networks at: http://www.slideshare.net/wildpackets/improving-the-ip-telephony-experience.

In the presentation, viewers will learn about how to test their network(s) before installing a VoIP system to identify and correct performance troubles as well as proactive troubleshooting techniques and using qualitative and quantitative analysis. Learn about ...

• The importance of Quality of Service (QoS) parameters to enable network devices to prioritize and give preference to packet streams that are sensitive to delay, packet loss, jitter, and other performance inhibitors.

• Network traffic pattern changes to look for that adversely affect VoIP

• The importance of providing alerts when VoIP performance declines

• Proactive troubleshooting tricks and tips that lead to a reliable VoIP implementation with first-rate voice quality.

If you are responsible for determining the health of your network before deploying VoIP, need to rid the network of VoIP-killing troubles before installation, or want to be equipped to assess the network when VoIP-killing troubles arise, you'll want to digest this slide presentation.