VoIP in Europe: Its bite is worse than its bark

news20090730-1New data from TeleGeography’s European VoIP & Triple Play Research Service reveal that voice over IP (VoIP) subscribers have grown from just over six million in 2005 to 34.6 million at year-end 2008. VoIP now accounts for more than 24% of fixed line telephone subscribers in Europe.

VoIP service revenues of EUR4.1 billion are still dwarfed by the nearly EUR36 billion generated by traditional switched fixed-line services. However, the impact of VoIP on the European fixed-line market is greater than its relatively modest subscriber and revenue share would suggest, due to the strong downward pressure VoIP-based competitors place on voice service prices.

Many European incumbent telephone companies have responded to this pressure by introducing discounted VoIP service, by slashing switched telephone service prices, and by marketing voice as one component of dual-play or triple-play bundles of voice, broadband, and video. These measures have helped incumbents to defend their market shares, but at the cost of sharply reduced voice revenues. Aggregate revenues from switched and VoIP telephone service have fallen from EUR49.4 billion in 2005 to EUR39.9 billion in 2008 and are projected to decline to only EUR26.2 billion by 2013.

According to TeleGeography analyst Paul Brodsky, ‘Fixed-line telephony was the cash cow that allowed incumbents to invest in mobile telephony, broadband, and video services. However, in Europe today, voice is increasingly just a loss leader, used to sell broadband and video services.’

Monitoring IP Telephony/Unified Communication

Many business start with VoIP as the first application to start there unified communication strategy. Once you have deployed unified communication, you need to continuously monitor and manage the environment. This will help you ensure agreed-upon service levels, including the availability of service and the quality of end user experience. You need

  • Visibility into all the elements of unified communication, from infrastructure to data applications and IP telephony
  • Assess Voice quality in real time
  •  Measure the end-user experience of applications
  • Reduce the time to resolve incidents

Why Monitor Your VoIP Traffic?

  • Any changes to the underlying LAN/WAN infrastructure will now affect VoIP Performance
  • Traditional Telephony (PSTN) each call is carried on a secured fixed bandwidth circuit for carrying the voice.
  • Packet Switched networks are designed to carry data, which are not time-sensitive.
  • With IP Telephony Voice is shared with Data as well as other voice calls over the same network
  •  IP networks allow each packet to independently find the most efficient path to the intended destination.
  •  Voice could arrive with different end-to-end delays, arrive out of sequence, or possibly not arrive at all.

Primary Challenge of Managing Unified Communication

Maintaining the quality and consistency of voice conversation in an ever-changing network environment.

Learn more here

Network Topology – Spend Less Time Documenting Your IT Network and More Time Managing It

Reasons for Understanding your Network Topology

Quickly reduce the area for fault detection

Reduce confusion when a fault occurs

Reduce time and effort when it comes time for Network expansion

Four Simple Guidelines for network documentation

Relevance – Avoid the temptation of capturing everything, if you collect to much information then it will not get updated, and then it becomes obsolete.

Easy – Use the KISS approach and make is simple.  Using an automated system allows you to easily update the network diagram and keep your inventory up to date.

Current – if the information is not current it only takes a couple of instances where a technician refers to this document to give the impression this is not important.  You need to automate this process so that the information being used is always current.

Safe  – Once compiled this information is sensitive and can contain, device passwords, SNMP community strings.  So you need access control to this information.  Your information should itself be password protected and encrypted whenever possible.  Your system needs to be available, even the best documentation scheme can’t help if it isn’t available when it’s needed.  Plan to have the documentation available when an outage occurs.  One solution is to set its own system up, that is independent of the network, which is then available at all times.

More information on what can help you develop your network topology here

What you don’t know WILL hurt you!

Why do contact center managers test their systems?

  • So they don’t get fired when it all goes sideways 23 minutes into peak busy hour?
  • So their customers won’t be inconvenienced when they use that new self-service, speech reco- enabled, web services-fed hosted IP voice portal
  • So their customers won’t be frustrated when they opt-out and get CTI-MPLS-transferred to Krissy in Bangalore or dropped?
  • Maybe all of the above?…

Everyone knows a well-crafted contact center solution is a thing of beauty.  But it is also complex, kind of like Monster Trucks – designed to perform spectacular feats in a really cool way but most importantly to get the crowd to say “WOW! That was awesome! Let’s do it again!” Making sure the “Wow!” is really there is a big part of contact center planning these days.

Everyone knows they don’t know if the “Wow!” is really there. They know they won’t know whether or not their systems have been properly implemented end-to-end until they turn them on and take them out for a run. And they certainly would prefer the maiden voyage not be with live customers whose first use becomes their last use when the “Wow!” turns to “Whoa!” Those customers decide quickly that from now on they might as well 0-out & talk to Krissy in the first place.

But no one knows what they don’t know.

And that’s the real reason they test. The smart ones know there are things they don’t know they don’t know. But they know they want to know.

It is interesting what we have discovered when running a test and I cannot count the number of times I have personally heard “If I hadn’t heard it I wouldn’t believe it – I had not idea!  But how would you know unless you test and monitor your system.

Let us help you Go Live with Confidence, with our test systems.  Find out more about all our test systems here

http://www.telnetnetworks.ca/en/technology-solutions/contact-centre-testing.html

Have a great day!

Does Adding Bandwidth Solve all VoIP Problems

The amount of bandwidth required by a VoIP call can be deceptive. The codecs send data at relatively slow data rates. G.711 typically consumes the most bandwidth, operating at 64 kbps. But for every VoIP packet that is sent, an RTP, UDP, and IP header are added. These headers add 12, 8, and 20 bytes respectively. So while G.711 sends data payloads at 64 kbps, the actual amount of bandwidth required is more like 87 kbps due to the additional headers.

Knowing the required bandwidth is important, and ensuring that you have the right link in place to support the maximum number of calls that you need to support.   By completing a total VoIP Assessment of your network you can then understand your network typical usage patterns, topology, and device configuration before you deploy VoIP on the network.

It is also important to understand that adding more bandwidth does not solve all problems.   Although Bandwidth can reduce congestion which would enable those links to support more VoIP calls with higher call Quality, it will not reduce delay or latency.  VoIP traffic is extremely sensitive to delay.  Slowing down a VoIP packet by more than 150 milliseconds can cause serious quality problems. The delay encountered by a VoIP packet occurs in several areas:

Queuing – The time spent in router queues along the path through the network. The more congestion, the greater the queuing delay.

Serialization – The time it takes to put a packet on a network link interface. This number increases as the packet size increases and as the link speed decreases.

Propagation – The time it takes to travel from one point to another in the network. This time is a fixed value that’s directly related to geographical distance. A good rule of thumb is 10 microseconds per mile.

Jitter buffer – Each VoIP phone and gateway provides a jitter buffer to smooth out effects of network jitter (variations in delay among packets in the same stream). This buffer adds delay as the packet is held for playout.

The only latency component that might possibly be reduced by adding bandwidth is the queuing delay. The other latency components are not affected by additional bandwidth, so if you are experiencing call performance problems due to delay, adding bandwidth probably won’t help

By completing a full VoIP Assessment you can understand not only the traffic patterns that may cause congestion which would mean a that adding extra bandwidth will help, but also the delay and jitter components for proper call quality.  As networks are always changing you should proactively monitor call quality to ensure that you have trending information, which can tell you what upgrades will be necessary in the future and that the end user continues to have a great experience.

You can download a white paper on this subject here

Proactive Testing for Confident Customer Interactions

Don’t Let Your Customers Test Your Systems!

These systems are built on hardware and software infrastructures that are highly integrated, and for them to be successful the technology must be efficient and reliable in design and performance.   your customers expect that these systems work continuously and reliably to supply great service. Developing and maintaining successful contact center systems is a challenging task.  You may know that each component is working independently but how do you know that all the parts of your system are working together and creating a positive experience for your customers.

A key part in managing risk and achieving project objectives is testing the customer experience.  The importance of proactive testing during each phase of the implementation ensures all your systems are working as they should be.

By putting the proper load testing you can test your system just as your customers use the system and see if you can go LIVE with CONFIDENCE.

http://www.telnetnetworks.ca/en/technology-solutions/contact-centre-testing.html

 

VoIP Pre-Assessment

What do you need to do to complete a VoIP pre-assessment so that you know that everything is going to  work once you deplay VoIP onto your network.   There are 6 steps to completing a complete VoIP pre-assessment

Build a Network Inventory:  You need to understand what LAN/WAN components are going to affect your overall quality. Understanding the devices and the links that the voice will be traveling gives you a good understanding of the network infrastructure, and the path the voice will be travelling. This should be an automated process so that any changes made to the network are quickly understood.

Understand Current Network Utilization:   Once you understand the current network inventory, you should baseline the network for the current utilization on each device and link. For a more complete assessment all the applications consuming bandwidth should be identified along with answers to who, what, where, and when. This will identify all the business critical applications and their performance baselines in order to later evaluate the impact of change. You should understand the utilization over a period of time (minimum 7 days) to understand traffic and utilization patterns. The application data is readily available from devices supporting Internet Protocol Flow Information eXport (IPFIX) or Cisco IOS® NetFlow. The available bandwidth, network delay, and packet loss behavior can be determined. At the end of this segment you should understand the limitations of current network devices, and the potential for additional upgrades before VoIP deployment.

Configuration Assessment:   Voice traffic traverses your entire infrastructure including routers, switches, IP telephony servers, IP phones, and circuits. A recent industry report states that more than 50% of network failures are due to devices being improperly configured, either during initialization, or upgrades to compliance and policy revisions. Service quality can be disrupted if any of these components are malfunctioning or are mis-configured. This phase of the assessment allows you to compare information about device configuration gathered during the network inventory to a set of rules you supply, or vendor based recommendations, and reports on any disparities. This includes device OS revision, memory, CPU etc.

Model the Network: Based on the inventory and the current utilization information, you can then model the network to predict the call quality that can be expected based on the current network statistics. This information can be used to predict the VoIP quality for feasible CODECs. This increases deployment success through “what if” analysis. Upfront modeling narrows down the scenarios that you will want to emulate on a live network.

Synthetic Calls:  Predicts call quality, Although you may find that your overall network health is satisfactory to support voice you need to place synthetic calls between specific locations over the network and collect statistics over a representative time period. This gives you the metrics to provide you an accurate prediction of actual call quality that can be expected once VoIP is deployed. You may also want to add background traffic to simulate the impact of the addition of new application on the network or verify QoS configuration.

Reporting:  It is important to put all this information in a report, and be able to retrieve this as a base line. At anytime in the future you can compare this report with future addition and changes in the network.

For a the full white paper on this subject you can go here