VoIP and the move towards unified communications

Transcription

1 VoIP and the move towards unified communications Convergence of voice and data in the enterprise has become widely adopted as organisations move from a legacy environment with separate circuit-switched voice and packet-switched data to next generation solutions where voice and data co-exist on a single network through voice over IP (VoIP), and subsequently to unified communications. At the same time, the rapid growth of Bring Your Own Device (BYOD) provides benefits to both users and businesses but is placing increasing demands on the wireless infrastructure. Table of contents The scale of the challenge... 2 The principles of implementing VoIP... 2 Wireless VoIP and the ac standard... 3 Best practice VoIP implementation Performance management About Solutions from The combination of these two trends places increasing demands on both wired and wireless networks. It is creating increasing challenges for the network engineer, who has to ensure that users experience acceptable call quality whether they are using wired or wireless access while the organisation obtains the required return on their investment in a converged infrastructure. This White Paper looks at the move to VoIP and unified communications and how to address the resulting performance issues in wired and wireless networks for an increasingly mobile workforce, as well as the potential benefits of the ac standard.

2 The scale of the challenge There is considerable appetite to move away from the traditional PBX and copper-based networks to a single infrastructure that handles all communications. The use of VoIP in the corporate world is growing rapidly, driven by a combination of increasingly mature technology and a desire to reduce costs. More companies are being given access to superfast broadband connections, which make it much easier to host this type of telephony solution. Businesses using CAT5 or CAT6 cabling can now retire their telephone exchange and reduce the amount of cabling they have to manage by converging all communications onto a single infrastructure. In a survey of network engineers carried out earlier this year, 21% of respondents were already using VoIP and 15% were planning to extend or implement it in The rate of VoIP growth is such that in November 2013 AT&T asked the Federal Communications Commission in the US to plan for the retirement of traditional phone networks and transition to all-tcp/ip. The benefits of convergence are clear. By having a single network infrastructure, organisations can reduce their capital expenditure and create a more homogeneous environment that, in theory at least, should be easier to maintain. This should make monitoring and management more straightforward, improving performance and reducing MTTR and hence increasing user satisfaction. However, using the existing network to transport voice calls as well as data imposes a penalty in the amount of traffic the infrastructure can support. Now add to the equation increasing user demands for mobility and connectionless protocols, whether from company supplied laptops and phones or BYOD. Users expect the performance of mobile applications to keep pace with their wired counterparts. They want to use any device to access any application, anywhere, without lag or delay in their experience. When VoIP is introduced, more and more bandwidth hungry applications such as voice and video will be driven over the wireless infrastructure. 4G rollout will necessitate an unavoidable increase in the number of VoIP calls made from portable devices, as this is the only kind of communication that is offered by up to date networking technology. Most business WLANs will have been planned and sized to match the capacity and throughput demands of data traffic, and so will be unprepared for this increased load. The introduction of VoIP, and subsequently unified communications, will put a spotlight on wireless problems that may have been masked up to this point. This creates increased pressure to redesign and upgrade the wireless infrastructure to provide bandwidth at higher rates than ever before. The less obvious aspect to consider when implementing VoIP is the impact on users. Many will already understand the concept of VoIP through having made Skype calls. However, when making a business call, employees are used to picking up their desk phone and dialling, with guaranteed call quality. VoIP is a very different experience: there may be no handset; the user may find it difficult to hear the person at the other end of the call due to background noise; jitter, latency or packet loss may make the call incomprehensible or cause it to drop out completely. A VoIP implementation is likely to create many more calls to the Helpdesk as users struggle to achieve the same experience that they had with circuit switched calls. Problems will range from I can t get as many people onto a conference call as I used to to calls dropping off part-way through. If users find VoIP too difficult to use or regularly experience problems, they will find alternative solutions which will negate the cost savings and productivity improvements the organisation is trying to achieve. The principles of implementing VoIP As a result of all these factors, while the move to VoIP may reduce the amount of infrastructure needed, it will place increasing demands on both wired and wireless networks and create increasing challenges for the network engineer. Moving to VoIP is not a trivial endeavour, but yields great results when executed properly. Careful planning and the right tools are needed to avoid reducing business efficiency and ensure that the organisation achieves the required return on its VoIP investment. There are two key aspects to consider when planning a VoIP implementation. 1. It places increasing capacity demands on the network by increasing the amount of traffic to be supported. 2. The nature of packetised voice traffic is very different from that of the typical application transport, impacting on both voice quality and the use of the available bandwidth. 2

3 All packets are subject to latency, jitter and loss as they traverse the network from their source to their destination. Packet loss of more than 5% will start to affect voice quality. Data packets use TCP, which is connection oriented. If there is a delay, or receipt is not acknowledged, the protocol times out and data is resent. Hence the ramifications of these events are typically minor and often go unnoticed. In contrast VoIP utilises the UDP protocol, which is inherently connectionless. If a packet is lost, or delivery is taking too long, the sender of the data has no mechanism to resend or adjust the rate by which data is sent. As a result latency, jitter and packet loss can have a devastating effect on call quality, rendering conversations unintelligible and putting the investment in convergence at risk. 20ms ms Figure 1: Out of sequence packets: in the example above, there are two paths to the same destination. However, one path has a higher latency than the other, which can result in the packets arriving out of sequence at the destination. The inability of VoIP to adjust for network conditions also means that it uses whatever bandwidth is available. TCP can and will adjust, and so if VoIP is using a large proportion of the bandwidth TCP traffic will see low availability, slowing down applications. Adding a significant number of VoIP users can impact the utilisation of network segments, reducing both voice call quality and the speed at which standard TCP applications perform. To address this problem and preserve the integrity of calls requires two different classes of service. Most networks use QoS technologies to protect and prioritize their VoIP traffic by tagging traffic at the device level with a queue marker (a Differentiated Service Code Point or DSCP) and setting parameters for how devices in their network treat such traffic. It is usually allowed top priority in being forwarded through the device, as well as some type of rate limit to ensure that data applications continue to perform at the levels users expect and demand. Wireless VoIP and the ac standard The difference between voice and data traffic becomes even more noticeable in a WLAN. When implementing VoIP in wireless networks, collisions in the wireless link will reduce capacity and increase latency and jitter. The standard has a built in algorithm to reduce collisions and hence lessen these problems, but this creates an additional overhead on each packet which will reduce the bandwidth available. If WLAN bandwidth is already at its limits, it may not be possible to add VoIP without increasing bandwidth. It is also important to ensure that the service provider can deliver the required level of service. If bandwidth is a concern, another option is to increase the compression of the audio signal, but the trade-off is that any packet loss will have a much more significant impact on call quality. Increased use of BYOD and growing demand for online video which, like voice, is time sensitive are also resulting in increasing congestion in the 2.4GHz spectrum. Users are comfortable with wireless they like the freedom and flexibility it provides, whether they are in the office, working from home or on the move. This congestion leads to interference and has a detrimental effect on the user experience ac is an IEEE draft amendment to the wireless protocol which attempts to address the congestion issue by moving wireless traffic to the 5GHz band. At the moment this band is less busy, so the move should provide significant benefits to users. The protocol is still at the draft stage, although ac routers are now available. It is expected to be ratified in February 2014 and will be backwards compatible with n. 3

4 802.11ac is also known as the very high throughput (VHT) amendment and claims to be designed to provide faster throughput at greater distance, improving the spectral efficiency of Wi-Fi and building on the improvements made with n. It does this through two features. First, it offers wider channels, which will support higher data rates, although this means that there will be fewer available channels in the 5GHz band compared with 2.4GHz. Second, it uses multiuser Multiple Input Multiple Output (MIMO) to support simultaneous transmissions to multiple clients, which maximizes RF band utilisation. It can handle up to 4 spatial streams per client, and each device continuously receives the full bandwidth on offer, which should speed up data transfer and improve the user experience. In contrast, an n access point can only send and receive data from one device at a time. A comparison between the two standards is shown in Table n ac Frequency band 2.4GHz and 5GHz 5GHz Channel width 20 and 40MHz 20, 40, 60, 80MHz (option 160MHz) Spatial streams 1 to 4 1 to 8 (up to 4 per client) Multiple user MIMO No Yes Single stream max. client data rate 150Mb/s 433MB/s (if 80MHz channel) Table 1: Comparison of n and ac protocols In the short term ac will not solve the congestion issue, as the majority of devices will still be contending for the 2.4GHz band. In the medium term, if the organisation makes a strategic decision to implement ac and as 5GHz devices come into use, the impact will become apparent. This will eventually lead to 5GHz congestion, and so the next step may be the introduction of dual standard devices, and/or the use of the short range ad standard. It should be noted that deploying ac is more complex because there are fewer, wider channels, beam forming has to be considered and there are regulatory variances, making RF management and control essential. Best practice VoIP implementation As we have seen, visibility into the performance of both voice and data traffic in a unified way is needed to deliver the required end user experience, and proper planning and design is vital. The key to successfully managing a VoIP implementation is a management solution with the breadth and depth to enable the benefits of VoIP to be delivered and the complexities mitigated. We recommend a five step process to implement VoIP. Ensure Infrastructure is VoIP ready Verify all elements communicate properly Test performance between sites and stress test Monitor continuously and optimise performance Educate users 1. Ensure that the infrastructure is VoIP ready by carrying out a complete assessment of the network, from LANs and WANs to desktops and phones. Baseline and monitor actual usage and performance to determine if the data network needs to be modified to handle voice calls. If bandwidth is already at its limits in certain parts of the network, for example, this may need to be increased to implement VoIP successfully. Bandwidth priority will need to be given to real time applications not only VoIP but also videoconferencing/streaming and these will require careful QoS allocation. When reviewing the WLAN, the White Paper BYOD Without Tears provides further information on planning and deploying BYOD which may be helpful. 2. Verify that all elements continue to communicate properly during the implementation, from jacks, cables and phones to switches and servers. Establish a performance baseline by generating synthetic VoIP traffic to confirm quality prior to full deployment. As the implementation progresses new issues may arise, such as new sources of interference in the WLAN, which will need attention. 4

5 3. Carry out a stress test when deployment is completed, looking at both overall performance and performance between specific sites. This will enable any potential bottlenecks to be identified and preventive action taken before they have a significant impact on users. 4. Actively monitor VoIP performance from various points on the network, including the core, individual routers and WAN links. VoIP problems can have a myriad of causes, from physical problems on the local loop and over utilised ports to misconfigured class of service settings and high levels of jitter within the voice application. Addressing this requires a broad management and troubleshooting strategy for the converged network. Real-time and back in time perspectives are vital to proactively circumvent emerging situations resolve ongoing anomalies as rapidly as possible, identify and correct intermittent problems, and optimise performance. 5. Educate users about VoIP. As was discussed earlier, making a VoIP call is very different from a traditional phone call, and users need to know what to expect and what to do if they experience problems. Some unified communications systems have an indicator of call quality, but in our experience these can be misleading. What a user may think is poor quality of service could be a software fault or some other issue. Performance management To achieve user acceptance, VoIP must provide high quality voice services without compromising the performance of data applications. This requires time correlated performance management data across traffic makeup, application performance, and VoIP stream quality by site, with the granularity to examine individual calls. Simultaneous network performance testing at various points on the network, combined with end to end graphical path analysis, will help to quickly identify the source of the problem. Real-time and back-in-time perspectives will enable the network engineer to resolve issues quickly, identify and correct intermittent problems and spot emerging situations and address them before they affect users. The best solutions integrate diagnostic data from the wired and wireless aspects of the network in a single interface and can quickly identify whether the problem is linked to the WLAN or somewhere else in the network. Since most IT staff work in silos and are specialised in one specific area, having a solution that is aware of all three provides a collaboration point that will help IT organisations understand and resolve the problem faster, leading to better collaboration, quicker MTTR and a higher quality of end user experience. A performance management solution will help to provide the answers to questions such as Is the network highly utilised and, if so, who is utilising it? Is the network healthy? Are key links performing properly at Layer 1? Which site has the worst VoIP performance and does that correlate with application degradations? Problem solving will usually begin with an alert about poor performance at a particular site. The first stage is to look at the network level and carry out traffic analysis. It is then possible to systematically drill down to the specific site and compare call quality between users to see if the problem is occurring for all users at the site or if the quality problem is limited to a specific call or calls. By capturing conversation flow and the associated packets, call statistics can be viewed and the call replayed to assess the problem, with a visual representation of traffic flow and quality. For wireless calls, an analysis tool will monitor the quality of each call and give it a WiR value and WiMOS score, looking at both sides of the call (i.e. from phone to access point and vice versa). A visual, colour coded representation will quickly show which calls have problems, and historic data will enable identification of problems tied to a particular phone, channel or time period. Further study of the performance metrics will then help to identify root cause. Reasons for changes in quality on an individual call can be examined, such as changes in packet jitter, signal quality or a spike in the number of users using the same access point. 5

6 ms 5ms 10ms 15ms 20ms 25ms 30ms Figure 2. Jitter distribution from VoIP report. The metrics enable a review of the relative amounts of voice and data traffic to see how they are competing for channel resources. If applications were also running slowly at a site which is experiencing VoIP problems, the next step is to examine CoS configurations to find out if VoIP has prioritisation and whether there is enough bandwidth at both ends. If there are applications in the voice queue, or the queue is misallocated in terms of the bandwidth or priority, it can create traffic contention, which will cause all the TCP applications to run into low traffic rate availability and start to slow down. Smooth roaming is another key requirement for wireless VoIP. If roaming isn t properly coordinated, or phones roam too often, users will experience dropped calls or other quality issues. An analyser will provide details of all roaming transactions, including possible reasons for the roaming instance, voice specific data such as signal strength leading up to the attempt, and packet transmission rates for the conversation. This will help to identify problem phones that may be constantly thrashing between access points or if there are issues which affect an entire channel and may be causing multiple phones to roam. Having identified root cause and made the appropriate changes, the monitoring system will show the impact of the changes on VoIP quality, traffic flow and application performance to validate that the fix has worked. If bandwidth is an issue, one option is to compress the VoIP. The trade-off is that any packet loss will have a much more significant impact on call quality. The performance management system can also be used proactively to optimise the converged network. This means: Capacity planning Traffic management Performance baselining, and... A commitment to continuous improvement. There are many things that can be done to adjust the interaction of VoIP with the infrastructure. Analysing management information across the network, from the edge to the core, will enable potential issues to be identified, such as a lack of bandwidth in specific sectors of the network. It will also determine whether specific applications such as file sharing or streaming media are impeding VoIP performance, and how traffic should be shaped to prioritise the services and applications that are most business critical and most sensitive to delays. We recommend carrying out traffic analysis before making any changes in order to verify that all traffic is business critical and set a baseline for performance. This may require policies to remove unwanted traffic before making any changes to call settings to optimise performance. Network performance measurements can also be used to carry out class of service testing and generate multiple test streams. This enables the engineer to validate SLAs and find out whether carrier links are supporting the agreed QoS. 6

7 About is the world-leading provider of network test and monitoring solutions to speed the deployment and improve the perfor mance of networks and applications. Leading enterprises and service providers trust products and expertise to help solve today s toughest issues and emerging challenges in WLAN security, mobility, unified communications and datacenters. Based in Everett, Washington, the company distributes products in more than 50 countries. For more information on our wireless solutions, visit Solutions from AirMagnet Wireless Solutions AirMagnet enables predictive modeling of enterprise WLANs, provides advice on AP placement and channel allocations, and runs what-if analyses on the impact of BYOD growth. After WLAN deployment, AirMagnet measures actual coverage and verifies true end-to-end performance and provides an Android app. to visualise coverage. More information at /wireless-resource-center OptiView XG Automated network and application analysis The OptiView XG is the first tablet specifically designed for the Network Engineer. It automates root-cause analysis of network and application problems allowing the user to spend less time on troubleshooting and more time on other initiatives. It is designed to support deployment of new technologies, including unified communications, virtualization, wireless and 10 Gbps Ethernet. The result is that new initiatives get up and running faster and network stay productive even in these days of smaller teams. More information at /xg OneTouch AT TM Client to Cloud troubleshooting in 60 seconds The OneTouch AT Network Assistant greatly reduces troubleshooting time through a streamlined, three-step approach: 1. The unique AutoTest replaces multiple tools and an hour of troubleshooting time. 2. A powerful set of network performance measurements to troubleshoot wired and Wi-Fi networks. 3. It enhances team collaboration through a simple web-remote interface and easy-to-use inline packet capture capabilities. More information at /OneTouchAT operates in more than 50 countries worldwide. To find your local office contact details, go to /contact Corporate Office: P.O. Box 777 Everett, WA USA info@flukenetworks.com European Office: P.O. Box 1550, 5602 BN Eindhoven Germany 0049-(0) France 0033-(0) UK 0044-(0) sales.core@flukenetworks.com 2013 Fluke Corporation All rights Reserved A 7