Tracking the Evolution of UC Technology

Tracking the Evolution of UC TechnologyDefining unified communications is more complicated than it seems, but a thorough understanding of UC technology is required before informed buying decisions can be made. Not only is the UC value proposition difficult to articulate, but it involves multiple decisions that impact both the IT group and end users.

In brief, UC is a platform that seamlessly integrates communications applications across multiple modes — such as voice, data and video — and delivers a consistent end-user experience across various networks and endpoints. While this describes UC’s technical capabilities, its business value is enabling collaboration, improving personal productivity and streamlining business processes.

At face value, this is a compelling value proposition, but UC offerings are not standardized and are constantly evolving. All vendors have similar core features involving telephony and conferencing, but their overall UC offerings vary widely with new capabilities added regularly.

No true precedent exists to mirror UC technology, which is still a fledgling service. The phone system, however, may be the closest comparison — a point reinforced by the fact that the leading UC vendors are telecom vendors.

But while telephony is a static technology, UC is fluid and may never become a finished product like an IP PBX. As such, to properly understand UC, businesses must abandon telecom-centric thinking and view UC as a new model for supporting all modes of communication.

UC technology blends telephony, collaboration, cloud

UC emerged from the features and limitations of legacy technology. Prior to VoIP, phone systems operated independently, running over a dedicated voice network. Using packet-switched technology, VoIP allowed voice to run on the LAN, sharing a common connection with other communications applications.

For the first time, telephony could be integrated with other modes, and this gave rise to unified messaging. This evolution was viewed as a major step forward by creating a common inbox where employees could monitor all modes of communications.

UC took this development further by allowing employees to work with all available modes of communication in real time. Rather than just retrieve messages in one place, employees can use UC technology to conference with others on the fly, share information and manage workflows — all from one screen. Regardless of how many applications a UC service supports, a key value driver is employees can work across different modes from various locations with many types of devices.

Today’s UC offerings cover a wide spectrum, so businesses need a clear set of objectives. In most cases, VoIP is already being used, and UC presents an opportunity to get more value from voice technology.

To derive that value, the spectrum of UC needs to be understood in two ways. First, think of UC as a communications service rather than a telephony service. VoIP will have more value as part of UC by embedding voice into other business applications and processes and not just serving as a telephony system. In this context, UC’s value is enabling new opportunities for richer communication rather than just being another platform for telephony.

Secondly, the UC spectrum enables both communication and collaboration. Most forms of everyday communication are one on one, and UC makes this easier by providing a common interface so users don’t have to switch applications to use multiple modes of communication. Collaboration takes this communication to another level when teams are involved.

A major inhibitor of group productivity has long been the difficulty of organizing and managing a meeting. UC removes these barriers and makes the collaboration process easier and more effective.

Finally, the spectrum of UC is defined by the deployment model. Initially, UC technology was premises-based because it was largely an extension of an enterprise’s on-location phone system. But as the cloud has gained prominence, UC vendors have developed hosted UC services — and this is quickly becoming their model of choice.

Most businesses, however, aren’t ready for a full-scale cloud deployment and are favoring a hybrid model where some elements remain on-premises while others are hosted. As such, UC vendors are trying to support the market with a range of deployment models — premises-based, hosted and hybrid.

How vendors sell UC technology

Since UC is not standardized, vendors sell it in different ways. Depending on the need, UC can be sold as a complete service that includes telephony. In other cases, the phone system is already in place, and UC is deployed as the overriding service with telephony attached. Most UC vendors are also providers of phone systems, so for them, integrating these elements is part of the value proposition.

These vendors, however, are not the only option for businesses. As cloud-based UC platforms mature, the telephony pedigree of a vendor becomes less critical.

Increasingly, service providers are offering hosted UC services under their own brand. Most providers cannot develop their own UC platforms, so they partner with others. Some providers partner with telecom vendors to use their UC platforms, but there is also a well-established cadre of third-party vendors with UC platforms developed specifically for carriers.

Regardless of who provides the platform, deploying UC is complex and usually beyond the capabilities of IT.

Most UC services are sold through channels rather than directly to the business. In this case, value-added resellers, systems integrators and telecom consultants play a key role, as they have expertise on both sides of the sale. They know the UC landscape, and this knowledge helps determine which vendor or service is right for the business and its IT environment. UC providers tend to have more success when selling through these channels.

Why businesses deploy UC services

On a basic level, businesses deploy UC because their phone systems aren’t delivering the value they used to. Telephony can be inefficient, as many calls end up in voicemail, and users waste a lot of time managing messages. For this reason, text-based modes such as chat and messaging are gaining favor, as is the general shift from fixed line to mobile options for voice.

Today, telephony is just one of many communication modes, and businesses are starting to see the value of UC technology as a way to integrate these modes into a singular environment.

The main modes of communication now are Web-based and mobile, and UC provides a platform to incorporate these with the more conventional modes of telephony. Intuitively, this is a better approach than leaving everyone to fend for themselves to make use of these tools. But the UC value proposition is still difficult to express.

UC is a productivity enabler — and that’s the strongest way to build a business case. However, productivity is difficult to measure, and this is a major challenge facing UC vendors. When deployed effectively, UC technology makes for shorter meetings, more efficient decisions, fewer errors and lower communication costs, among other benefits.

All businesses want these outcomes, but very few have metrics in place to gauge UC’s return on investment. Throughout the rest of this series, we will examine the most common use cases for UC adoption and explore the major criteria to consider when purchasing a UC product.

Thanks to Unified Communications for the article. 

 

How To Monitor VoIP Performance In The Real World

How To Monitor VoIP Performance In The Real WorldIt’s one of the most dreaded calls to get for an IT staff member – the one where a user complains about the quality of their VoIP call or video conference. The terms used to describe the problem are reminiscent of a person who brings their car in for service because of a strange sound “ I hear a crackle”, or “it sounds like the other person is in a tunnel” or “I could only hear every other word – and then the call dropped”. None of these are good, and unfortunately, they are all very hard to diagnose.

As an IT professional, we are used to solving problems. We are comfortable in a binary world, something works or it doesn’t and when it doesn’t, we fix the issue so that it does. When a server or application is unavailable, we can usually diagnose and fix the issue and then it works again. But, with VoIP and Video, the situation is not so cut and dried. It’s rare that the phone doesn’t work at all – it usually “works” i.e the phone can make and receive calls, but often the problems are more nuanced; the user is unhappy with the “experience” of the connection. It’s the difference between having a bad meal and the restaurant being closed.

In the world of VoIP, this situation has even been mathematically described (leave it to engineers to come up with a formula). It is called a Mean Opinion Score (MOS) and is used to provide a data point which represents how a user “feels” about the quality of a call. The rating system looks like this:

How To Monitor VoIP Performance In The Real World

Today, the MOS score is accepted as the main standard by which the quality of VoIP calls are measured. There are conditional factors that go into what makes an “OK” MOS score which take into account (among other things) the CODEC which is used in the call. As a rule of thumb, any MOS score below ~3.7 is considered a problem worth investigating, and anything consistently below 2.0 is real issue. *(many organizations use a different # other than 3.7, but it is usually pretty close to this). The main factors which go into generating this score come from 3 KPI’s

  1. Loss
  2. Jitter
  3. Latency / Delay

So, in order to try and bring some rigor to monitoring VoIP quality on a network (and get to the issues before the users get to you) network staff need to monitor the MOS score for VoIP calls. In the real world there are at least three (separate) ways of doing this:

1) The “ACTUAL” MOS score from live calls based on reports from the VoIP endpoints

Some VoIP phones will actually perform measurements of the critical KPI’s (Loss, Jitter, and Latency) and send reports of the call quality to a Call Manager or other server. Most commonly this information is transmitted using the Real Time Control Protocol (RTCP) and may also include RTCP XR (for eXtended Report) data which can provide additional information like Signal to Noise Ratio and Echo level. Support for RTCP / RTCP XR is highly dependent on the phone system being used and in particular the handset being used. If your VoIP system does support RTCP / RTCP XR you will still need a method of capturing and reporting / alarming on the data provided.

2) The “PREDICTED” MOS score based on network quality metrics from a synthetic test call.

Instead of waiting for the phones to tell you there is a problem, many network managers implement a testing system which makes periodic synthetic calls on the network and then gathers the KPI’s from those calls. Generally, this type of testing takes place completely outside of the VoIP phone system and uses vendor software to replicate an endpoint. The software is installed at critical ends of a test path and then the endpoints “call” each other (by sending an RTP stream from one endpoint to another). These systems should be able to exactly mimic the live VoIP system in terms of CODEC used and QoS tagging etc so that the test frames are passed through the network in exactly the same way that a “real” VoIP call would be. These systems can then “predict” the Quality of experience that the network is providing at that moment.

3) The “ACTUAL” MOS score bases on a passive analysis of the live packets on the network.

This is where a passive “probe” product is put into the network and “sniffs” the VoIP calls. It can then inspect that traffic and create a MOS score or other metrics which is useful to determine the current quality of service being experienced by users. This method removes any need for support from the VoIP system and also does not require the network to handle additional test data, but does have some drawbacks as this method can be expensive and may have trouble accurately reading any encrypted VoIP traffic.

Which is best? Well, they both all have their place, and in a perfect world an IT staff would have access to live data and test data in order to troubleshoot an issue. In an even more perfect world, they would be able to correlate that data in real time to other potentially performance impacting information like router / switch performance data and network bandwidth usage (especially on WAN circuits).

In the end, VoIP performance monitoring comes down to having access to all of the critical KPI’s that could be used to diagnose issues and (hopefully) stop users from making that dreaded service call.

How To Monitor VoIP Performance In The Real World

Thanks to NMSaaS for the article.

Can Your Analyzer Handle a VoIP Upgrade?

Is your old VoIP or PBX system rapidly approaching the end of its life? Your network has changed substantially since its deployment many moons ago, making this an ideal time to investigate new VoIP systems and ensure your existing monitoring solution can keep pace with the upgrade.

Here are 4 critical areas for consideration to determine whether your monitoring tools can keep pace with the new demands of a VoIP upgrade.

1. SUPPORTING MORE THAN ONE IT TEAM:

If you’re shifting from a traditional PBX system to a VoIP solution, chances are the system will be managed by more than one team. While you might live and breathe packet-level details, the voice team is accustomed to metrics like jitter, R-Factor, and MOS.

CAN YOUR MONITORING SOLUTION PROVIDE VOIP-SPECIFIC QUALITY ASSESSMENTS PLUS PACKET AND TRANSACTION DETAILS FOR PROBLEM RESOLUTION?

Can Your Analyzer Handle a VoIP Upgrade?

2. ADDRESSING CONFIGURATION CHALLENGES:

In rolling out large VoIP deployment systems, device and system misconfigurations can get the best of even the most experienced network team. Have you hired VoIP consultants or is this a DIY project? If the ball’s in your court to bring VoIP to the desktop, be sure to run through a pre-deployment and monitoring capabilities checklist to ensure for successful implementation.

3. ISOLATING THE ROOT CAUSE:

Have you ever seen users or departments experiencing bad MOS scores only to ask yourself, “Now what?” How do you quickly navigate to the source of the problem?

It’s more than exonerating the network.

Your solution should let you isolate the source of quality problems. Does your solution allow you to determine whether the call manager or a bad handset might be at the root of your VoIP frustrations?

4. SUPPORTING MULTI-VENDOR INSTALLATIONS:

Many larger IT environments are now implementing VoIP solutions from multiple vendors. For example, you’ve already rolled out Cisco® to the desktop, and have been tasked to deploy Avaya® to the call center. Does your analyzer provide detailed tracking for multiple vendors? Has your monitoring solution been configured to understand the differences in how each VoIP system handles calls? Without this support, you may be forced to toggle between multiple screens to troubleshoot or reconcile various quality metrics to assess VoIP performance.

Can Your Analyzer Handle a VoIP Upgrade?

CONCLUSION

Understanding the changes in the environment, ensuring rapid problem isolation, tackling potential configuration challenges, and assessing your solution’s support for multiple vendors are the keys to ensuring a successful rollout.

Thanks to Network Instruments for the article.

End User Experience Testing Made Easier with NMSaaS

End user experience & QoS are consistently ranked at the top of priorities for Network Management teams today. According to research over 60% of companies today say that VoIP is present in a significant amount of their networks, this is the same case with streaming media within the organization.

End User Experience Testing Made Easier with NMSaaS

As you can see having effective end user experience testing is vital to any business. If you have a service model, whether you’re an actual service provider like a 3rd party or you’re a corporation where your IT acts as a service provider you have a certain goal. This goal is to provide assured applications/services to your customers at the highest standard possible.

The success of your business is based upon your ability to deliver effective end user experience. How many times have you been working with a business and have been told to wait because the businesses computers systems were “slow”. It is something which we all have become frustrared with in the past.

End User Experience Testing Made Easier with NMSaaS

To ensure that your organization can provide effective and successful end user experience you need to be able to proactively test your live environment and be alerted to issues in real time.

This is comprised of 5 key elements:

1) Must be able to test from end-to-end

2) Point to Point or Meshed testing

3) Real traffic and “live” test, not just “ping” and trace route

4) Must be able to simulate the live environments

  • Class of service
  • Number of simultaneous tests
  • Codecs
  • Synthetic login/query

5) Must be cost effective and easy to deploy.

NMSaaS is able to provide all of these service at a cost effective price.

If this is something you might be interested in, or if you would like to find more about our services and solutions – why not start a free 30 day trial today?

End User Experience Testing Made Easier with NMSaaS

Thanks to NMSaaS for the article.

The Advancements of VoIP Quality Testing

The Advancements of VoIP Quality TestingIndustry Analysts say that approximately 85% of today’s networks will require voip quality testing upgrades to their data networks to properly support high-quality VoIP and video traffic.

Organizations are always looking for a way to reduce costs, and that’s why they often try to deploy VoIP by switching voice traffic over to a LAN or WAN links.

In a lot of cases the data networks which the business has chosen handle VoIP traffic accordingly, generally speaking voice traffic is uniquely time sensitive, it cannot be qued and if data grams are lost the conversation can become choppy.

To ensure this doesn’t happen many organizations will conduct a VoIP quality test in the pre and post deplomyent stage.

Pre Deployment testing

There are several steps network engineers can take to ensure VoIP technology can meet expectations. Pre-deployment testing is the first step towards ensuring the network is ready to handle the VoIP traffic.

After the testing process, IT staff should be able to:

  • Determine the total VoIP traffic the network can handle without audio deprivation.
  • Discover any configuration errors with the network and VoIP equipment.
  • Identify and resolve erratic problems that affect network and application performance.
  • Identify security holes that allow malicious eavesdropping or denial of service.
  • Guarantee call quality matches user expectations.

Post deployment testing

Places that already have VoIP/video need to constantly and easily monitor the quality of those links to ensure good quality of service. Just because it was fine when you first installed it, doesn’t mean that it is still working well today, or will be tomorrow.

The main objective of post deployment VoIP testing is to measure the quality and standard of the system before you decide to go live with it. This will in return stop people from complaining about poor quality calls.

Post-deployment testing should be done early and often to minimize the cost of fault resolution and also to provide an opportunity to apply lessons learned later on during the installation.

In both pre and post deployment the testing needs to be simple to setup and provide at a glance actionable information including alarms when there is a problem.

Continuous monitoring

In many cases your network changes every day as devices are added or removed, these could include laptops, IP phones or even routers. All of these contribute to the continuous churn of the IP network experience.

A key driving factor for any business is finding any faults before they become a potential hindrance on the company, regular monitoring will eliminate any potential threats.

In this manner, you’ll receive maximum benefit from your VoIP investment. Regular monitoring builds upon all the assessments and testing performed in support of a deployment. You continue to verify key quality metrics of all the devices and the overall IP network health.

If you found this interesting have a look at the recording of one our webinars on this topic, you will get an in-depth look on this topic.

The Advancements of VoIP Quality TestingThanks to NMSaaS for the article.

Three VoIP Quality Killers

VoIPManaging VoIP quality is a constant headache for engineers. Here are some ways to avoid some common problems.

While VoIP has been considered mainstream in the enterprise for the past couple of years, network engineers still face a challenge to ensure consistent VoIP quality. Unplanned or surprise changes to the network can leave end users unhappy and administrators scratching their heads. Even the smallest configuration and setting changes have the potential to immediately and adversely impact applications. One way to guard against performance issues is to use retrospective analysis — long-term packet capture for troubleshooting sporadic network issues — to identify and fix common VoIP quality problems more quickly. Here are some steps to follow:

Assess quality of service. Quality of service (QoS) settings are often the first step to manage VoIP. If QoS settings are not set correctly, the VoIP call doesn’t get passed at the priority it should. That can really affect call quality. Since network devices take care of setting the priority, incorrect QoS tagging can cause the quality of the calls to deteriorate. Customers will call in to figure out why they have one side of a phone conversation that sounds perfect and has a good mean opinion score while the other side is bad. Not having QoS tags properly configured can cause backups in the VoIP traffic, which can cause high jitter. If packets aren’t arriving when they’re supposed to arrive, there will be delays and the call will start sounding choppy.

To pinpoint the source of the problem, it is often necessary to begin at the packet level, using a retrospective network analyzer. A network recorder or advanced probe appliances placed at different points on the network, such as in front of different switches or routers, can collect a massive amount of data. Using different tools within the network analysis application, you can assess the QoS tag settings to determine if they are the same on both ends. If they’re not, this method can help you identify the offending hardware device. If QoS settings have already been checked and there are still VoIP quality issues, there are a few other culprits that are likely to interfere with service delivery.

Changing codecs. The next item on the list is to check whether codecs are changing. When calls are established, phone systems communicate which codecs are supported. Typically, phones will use the highest-quality codec when the call starts. But if something happens during the call, and the phone senses high jitter or data loss, the codecs will get changed. By looking at which codecs were used inside a Real-Time Transport Protocol stream, you can actually see, for example, that a call may have started off at the G.711 codec, which is 64 Kbps sampling, then changed to G.729, which is only 8 Kbps sampling. This makes for a less choppy-sounding audio, but causes a drastic drop in quality. Typically, codec changes happen due to network congestion.

High network utilization and network congestion. Another way to reduce the time it takes to troubleshoot VoIP issues is to establish accurate baselines. In larger enterprise environments, this can mean several network probes collecting data. In other words, what is “normal” on the network? What is the normal utilization? What are the normal protocols that are running? Baselines let you know when things get out of whack, because network utilization can affect VoIP quality.

While monitoring packets, it is important you pay close attention to network use. Tracking timestamps during instances of high jitter or packet loss and comparing this data to the intervals of higher network traffic can provide valuable insights.

One enterprise VoIP user, the global finance company Santander, reported solving a VoIP quality issue in this way: An Adobe update to Acrobat caused significant problems to the company’s network — calls suddenly began to drop off across its call centers. By looking at the high spike in traffic compared to when calls started dropping, network administrators were able to identify high utilization as the cause of the problem. They then evaluated post-event capture data and discovered a surge of Internet traffic flowing to Adobe IP addresses. The fix, as it turned out, was rather simple: Santander IT set up Adobe Reader to access local servers for updates instead of going out to the Internet. Problem solved.

By following these steps to avoid VoIP quality killers, troubleshooting time can be reduced and service delivery improved. A retrospective network analyzer can also help engineers spot future trends and allow them to resolve issues before they become problems.

Thanks to Tech Target for the article.

Webinar- The Importance of VoIP Testing

Industry Analysts say that approximately 85% of today’s networks will require upgrades to their data networks to properly support high-quality VoIP and video traffic.

Organizations are always looking for a way to reduce costs, and that’s why they often try to deploy VoIP by switching voice traffic over to a LAN or WAN links.

In a lot of cases the data networks which the business has chosen cannot handle VoIP traffic accordingly, generally speaking voice traffic is uniquely time sensitive, it cannot be qued and if data grams are lost the conversation can become choppy.

To ensure this doesn’t happen many organizations will conduct a VoIP quality test in the pre and post deplomyent stage.

NMSaaS Free Webinar - The Importance of VoIP Testing

Pre-Deployment testing

There are several steps network engineers can take to ensure VoIP technology can meet expectations. Pre-deployment testing is the first step towards ensuring the network is ready to handle the VoIP traffic.

After the testing process, IT staff should be able to:

  • Determine the total VoIP traffic the network can handle without audio deprivation.
  • Discover any configuration errors with the network and VoIP equipment.
  • Identify and resolve erratic problems that affect network and application performance.
  • Identify security holes that allow malicious eavesdropping or denial of service.
  • Guarantee call quality matches user expectations.

Post deployment testing

Places that already have VoIP/video need to constantly and easily monitor the quality of those links to ensure good quality of service. Just because it was fine when you first installed it, doesn’t mean that it is still working well today, or will be tomorrow.

The main objective of post deployment VoIP testing is to measure the quality and standard of the system before you decide to go live with it. This will in return stop people from complaining about poor quality calls.

Post-deployment testing should be done early and often to minimize the cost of fault resolution and also to provide an opportunity to apply lessons learned later on during the installation.

In both pre and post deployment the testing needs to be simple to setup and provide at a glance actionable information including alarms when there is a problem.

Continuous monitoring

In many cases your network changes every day as devices are added or removed, these could include laptops, IP phones or even routers. All of these contribute to the continuous churn of the IP network experience.

A key driving factor for any business is finding any faults before they become a potential hindrance on the company, regular monitoring will eliminate any potential threats.

In this manner, you’ll receive maximum benefit from your VoIP investment. Regular monitoring builds upon all the assessments and testing performed in support of a deployment. You continue to verify key quality metrics of all the devices and the overall IP network health.

If you missed our webinar last week you can watch the live recording of it here

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Thanks to NMSaaS for the article.