HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE

HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE

Introduction Expanding numbers of SMBs are now relying on the cloud for communication and other mission-critical services. In the case of Hosted PBX service, migrating to the cloud allows SMBs to use their existing internet connectivity (referred to as BYOB or bring your own bandwidth ) - an option that greatly increases deployment possibilities at the same time that it reduces cost. The promise of a lower Total Cost of Ownership, as well as cutting edge features and productivity tools, without having to maintain an on-premise system, is an attractive offer. However, resellers are finding out that is often difficult to meet the customer s Quality of Service expectations when a Hosted PBX is being delivered as an Over-The-Top (OTT) service. Quality of Service issues can be caused by latency, jitter or network congestion within the LAN of the client, at the demark between the LAN and the Internet or within the ISP network. Even when the problems are shown to be outside of the resellers infrastructure, it is often difficult to convince the client that these issues are beyond the control of the VoIP Reseller. Therefore, pinpointing and helping to correct these issues are often the difference between keeping a customer and losing them to the next provider that assures the client that their current reseller does not provide a good service and that they can solve all of these issues with their superior product. This whitepaper discussed the issues that need to be researched and addressed so that a reseller can understand the factors that affect the Quality of Service that is delivered to their Hosted PBX client base. HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 2 11

What is Quality of Service The quality of a connection is often a difficult concept to understand, as it can be affected by a number of factors. The actual speed of the connection, throughput, error rate, and the priority that any packet is given can all impact network performance; this is generally interpreted as the quality of the connection. Quality is crucial in the case of VoIP and other time-sensitive applications. From a technical perspective good voice quality involves minimizing interruptions ( blips ) in the communication stream. These interruptions are caused by Packet Loss, Packet Delay (sometimes called latency) and Packet Delay Variance (sometimes referred to as simply variance, or jitter). Packet Loss Packet loss occurs when one or more packets travelling across a network fail to reach their destination. Packet loss can be caused by signal degradation over the network medium due to multi-path fading, packet drop because of channel congestion, corrupted packets rejected in-transit, faulty networking hardware, faulty network drivers or normal routing routines. Packet loss is expressed as a percentage of the total number of packets sent. For acceptable voice quality, packet loss should be less than 10 percent. Packet Delay Packet Delay refers to the amount of time it takes for a packet to be transmitted across a network from source to destination. For voice communication, high delay can be disruptive because it impedes the synchronicity of a conversation, often causing speakers to step on one another. While a small amount of delay is unavoidable in packet networks, problems can occur when the delay is not uniform from packet to packet, a condition referred to as packet delay variance or jitter. Packet delay is expressed in milliseconds. For acceptable voice quality, packet delay should be below 120 milliseconds (120ms). Packet Delay Variance Packet Delay Variance, or jitter, is the variation in packet delay caused by queuing, contention and serialization effects on the path through the network. In general, higher levels of packet delay are more likely to occur on either slow or heavily congested links. Packet Delay Variance is expressed in milliseconds. For acceptable voice quality, packet delay should be below 10 milliseconds (10ms). HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 3 11

Connectivity The primary way to address Quality of Service issues is to ensure the integrity of the customer s connectivity. A failure of an SMB's connection to the Internet can cripple a company that has moved mission-critical services, such as Hosted PBX, to the Cloud. The failure to connect to the Internet can be caused by a variety of reasons. The physical connection can be disrupted or there can be a logical or protocol issue that creates a failure condition even if there is a physical connection The failure can be caused by a local problem or an issue deep in a provider s network. Redundancy The reliance on one connection to an ISP or the reliance on one ISP is the most common single point-of-failure in a cloud architecture. Multiple Connections to a Single ISP One solution to this issue is to have multiple connections to the ISP. This involves more than just ordering multiple IP addresses or multiple drops from the ISP. This is only effective if there are multiple physical routes into the building or facility. If all connections from the ISP come through the same conduit, then a single physical cut can break all those connections. This is true even if one connection is a T1 and the second connection is DSL. But this solution could work to provide redundancy for logical issues in the ISP network. If two physical connections are not available, then it may be necessary to make the second connection to the ISP a wireless connection. Connections to Multiple ISPs Having multiple connections to different providers defends against an outage or interruption in the ISP network impacting the productivity of the SMB. With proper load balancing across two or more Internet connections, redundant and guaranteed internet connectivity can be achieved. Monitoring, Measuring, and Addressing HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 4 11

In order to get a handle on VoIP Quality of Service, it is important to understand the roles that monitoring, measuring and addressing play. Monitoring Monitoring refers to the activities related to gathering data that is relevant to VoIP Quality of Service so that the reseller can be alerted when a problem arises. When it comes to Hosted PBX, the goal in monitoring is always to be aware of a problem at its earliest detectable phase so that restorative actions can be taken as quickly as possible. Put another way, the goals of monitoring are to: Increase visibility and responsiveness to critical events, issues, or conditions Decrease average mean time to repair for problems or failures Minimize the impact on end users VoIP QoS monitoring requires the collection of data from the various points in the network that will be useful in determining when and where a trouble condition may occur. The most basic form of monitoring might involve running a continuous ping test between a Hosted PBX server and the WAN IP address of a SIP client, and highlighting when ping times exceed a certain level. This type of test is less than ideal. Although ping time can provide an approximation of delay, loss and variance, a good ping time reading is no guarantee that a user s perceived quality is acceptable. To be effective, a monitoring system must be able to monitor the actual delay, loss and jitter for every call. Further, because VoIP traffic flows synchronously in two directions, monitoring must be performed at both sides of the conversation in order to capture all the relevant data. Alerts A critical part of the monitoring activity is the ability to receive automated alerts when a metric that is being monitored exceeds a predetermined threshold. Measurement Measuring VoIP Quality of Service means tracking the delay, loss and variance of calls over time. A problem arises however, because the effect of these variables on the perceived quality of a call will differ depending on the codec that is used. Codecs The term codec refers to the sampling rate and compression scheme used to digitize an audio stream. As might be expected, codecs with higher fidelity require more bandwidth. In a VoIP call, the goal is to use the codec which results in the best possible user experience. But because packet loss and delay reduce voice quality, a trade-off must always be made between codec fidelity and bandwidth consumption. Some codecs were engineered for the highest audio HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 5 11

clarity possible. Others were designed to provide acceptable audio quality when faced with lossy connections. Although there are many codecs available to be used on Hosted PBX systems, most service providers consider G.711 and G.729 as the standards because virtually all of their existing vendors (SIP carriers and trunk providers) and equipment providers (SIP CPE and gateways) support these codecs. However, it is difficult to compare the quality of VoIP calls when the samples use different codecs. To counter this, VoIP Monitoring systems use the delay, loss and variance of a call to derive a Mean Opinion Score (MOS) as it applies to the codec that was used for a call. Mean Opinion Score (MOS) The MOS is a numerical indication of the perceived quality from the user's perspective and is expressed as a single number in the range between 1 and 5, where 1 is lowest perceived audio quality, and 5 is the highest perceived audio quality measurement. Using a MOS score allows the monitoring system to normalize scores regardless of the codec used. The table below shows the subjective interpretation of each MOS value. MOS Quality Impairment 5 Excellent Imperceptible 4 Good Perceptible but not annoying 3 Fair Slightly annoying 2 Poor Annoying 1 Bad Very annoying Generally speaking, the MOS for a VoIP call will never exceed 4.5 as that is the highest possible rating for a call using the G.711 codec. As a point of comparison, the highest possible rating for a call using the G.729 codec is 3.9. WAN Throughput WAN Throughput refers to the measurement of actual Internet bandwidth consumed relative to the amount of bandwidth available. Because the bandwidth available to BYOB clients often fluctuates, it is important to measure when usage approaches and/or exceeds the available bandwidth. Historical Reporting Key to any VoIP Monitoring system is the ability to view MOS and actual bandwidth measurements over an extended period of time. This is desireable because trends can be very helpful in troubleshooting problems. It is often possible to correlate a trend line with some other HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 6 11

factor that might account for reductions in the perceived quality of a call. Therefore, the ability to generate reports that illuminate these relationships should be considered mandatory. Addressing Quality of Service Issues The implementation of strategies discussed below can lessen the amounts of packet delay, loss and variance experienced by Hosted PBX end users. Improving Quality of Service Quite often, VoIP and Unified Communications is the first application that is moved to the cloud. However, many find out the hard way that VoIP does not always produce pristine calls when the available bandwidth is not effectively managed to ensure VoIP call quality. The bandwidth required by other applications and services can negatively impact voice quality. In a typical installation, when a single internet connection is utilized for both voice and data traffic, special measures must be employed to ensure good voice quality. In a typical situation, where the customer has a 4x1 DSL connection (that is 4Mbps downstream and 1Mbps upstream) a user that is uploading a file to a cloud storage service could send twenty 1500-byte data packets. If this payload is in the router s queue, followed by one time-sensitive voice packet, it will take roughly 300 ms to send those data packets to the cloud storage service over the 1 Mbps uplink. When the voice packet is finally sent, the resulting delay would result in degraded voice quality. Bandwidth Management Bandwidth Management is used by network administrators to minimize network congestion via traffic shaping techniques. There are several tools and techniques that can be encompassed by Bandwidth Management. Bandwidth Limits The most basic type of bandwidth management is bandwidth limiting, a technique by which the router will allocate a certain amount of bandwidth that any user or group of users can be using at one time. This is useful in preventing bandwidth starvation in the network when a handful of users utilize all the available bandwidth with media streaming applications, gaming, etc. The downside is that this type of control typically does not differentiate between voice traffic and other traffic, and therefore controls can not be applied granularly. HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 7 11

VoIP QoS Successful VoIP QoS strategies require the use of a router that is able to detect packets containing voice payload and then treat those packets differently from all other packets. Understanding all of the moving parts in a VoIP prioritization scheme can be difficult due to the myriad of settings and values that need to be managed in order to classify and prioritize voice traffic. Differentiation between voice and other data packets can be done in many ways, and routers employ various mechanisms. Typical methods include identifying voice traffic based on the the IP and/or MAC address of known voice endpoints, the SIP signalling ports being used, the range of media RTP ports, and class-of-service packet tagging mechanisms such DSCP and/or CoS, or VLAN tags. The most common way to manage VoIP QoS is to deploy a router that implements a one ahead bandwidth reservation algorithm. The QoS system should dynamically track all of the voice packets on the network and manage the traffic so there is always enough bandwidth reserved for one voice call more than the current number of active calls. In a very dynamic environment where traffic tends to spike, or there are many very short calls, the solution may need to keep enough bandwidth available for one, two or three more calls than are currently in progress in order to make sure that quality is never affected by these conditions. Upstream and downstream traffic present different problems, and are best addressed by different Quality of Service (QoS) mechanisms. Upstream traffic flows from the customer premise to the Internet, while downstream traffic flows from the Internet to the customer. Upstream QoS Introducing class-of-service in the packet layer addresses queuing delay by ensuring voice packets receive priority treatment - in much the same way that separate queues at airport check-in counters ensure priority service to first-class customers. By creating separate queues for Real-Time and Non-Real-Time traffic, it is possible to assign higher priority to the Real-Time queue and serve that traffic more quickly. Implementing a class-of-service approach in the upstream direction ensures that outbound voice packets get served before other packet types to prevent the Internet access link (the bottleneck) from becoming overloaded. Downstream QoS Implementing QoS in the downstream direction requires a different approach. Typically, customer-premise equipment (CPE) at user locations exerts no control over incoming traffic. For traffic flowing downstream from the Internet, the access router cannot control the volume of traffic that gets sent to it because users on the LAN can initiate file transfers or download email at will. The servers handling these requests may be located anyplace. Since the downstream rate normally cannot be controlled, the ISP s edge router commonly responds to overloading by discarding VoIP packets with the same frequency as any other packet type - resulting in packet loss to the VoIP session. Unlike other data traffic, which can be retransmitted so the impact on the user experience is simply slower service response, dropped VoIP packets will ruin a voice conversation. To resolve the problem of degraded voice quality for incoming traffic, the router at the customer premise must throttle back Non-RealTime traffic, to ensure voice packets are transmitted freely downstream. This can be thought of as creating a dynamic bottleneck for downstream traffic, HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 8 11

and is accomplished using the flow-control mechanisms that are part of the TCP-IP standard. Because 80% of Internet traffic is transported via TCP, Downstream QoS can be just as effective as packet prioritization for upstream traffic. The dynamic bottleneck adjusts to varying traffic patterns, such that whenever there are no VoIP calls in progress, the full downstream bandwidth is available for incoming data such as file downloads. Session Limit Session Limiting is another method for shaping traffic. Employing this method requires that the router identify voice sessions, and then actively block or reject sessions that will exceed the limit. This type of functionality is accomplished using a SIP application layer gateway (ALG), and can effectively control the total number of sessions that a user or group are allowed, either individually or together. Since routers can handle only a certain maximum amount of throughput, this method makes sure that a user or group of users does not overwhelm the router with too many sessions. This is most often the case with peer-to-peer applications or when a user is infected with malware. Content Filtering Bandwidth quality can also be affected when users attempt to move too much data over the available connection. This is often caused by users utilizing bandwidth for purposes other than work, such as streaming music or video, gaming, using social media, or inadvertently stumbling into malicious sites that intentionally overload connections. Content Filtering enables administrators to block access to these types of sites. SMBs can subscribe to online services that constantly monitor the Internet and can automatically block users from accessing these types of sites, thereby removing the bandwidth consumption that would otherwise occur. VLAN A Virtual LAN (VLAN) groups together a number of different devices so that they appear to be on the same physical network when they actually reside on different networks. When all devices are physically on the same network, network broadcasts are received by every device connected to that network. This can cause a decrease in the performance of the entire network. Grouping devices into smaller logical networks or VLANs enhances performance and increases security. Broadcasts are restricted to only those devices that are part of the VLAN. Network security policies can be tailored to job roles or departments even if users are in varied geographical locations and physical networks. This helps to reduce the cost of implementing and administering users and devices. In the QoS realm, VLANs are used to group SIP phones onto a single logical network to which packet prioritization techniques can be applied. Intelligent WAN Routing When multiple WAN connections are available, Intelligent WAN routing enables SMB sites to re-route traffic based on an ISP failure, latency, packet loss, and packet delay variance for each defined network route. Load Balancing is a variation on WAN routing that enables traffic to be spread across multiple connections to the ISP or across connections to multiple ISPs. With the proper route policy across the multiple connections, time-sensitive traffic such as VoIP can be separated from other network traffic and sent over the faster connection. HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 9 11

The QoS Device Without a way to quickly diagnose these local problems and help the client resolve them, users are quick to blame the Hosted PBX reseller and move on to another provider who promises that their service never has such issues. SkySwitch recommends that resellers deploy a quality of service device at the customer premise to overcome this problem. SimpleWan SimpleWan QoS Routers help resellers of Hosted PBX diagnose and correct many of the issues addressed above, all while ensuring that a secure and compliant environment is maintained. This can all be accomplished from a single web interface for all of a Reseller s clients, and requires minimal technical expertise on the part of the Resellers technical staff. SimpleWan is a subscription and hardware based firewall designed for small, medium and enterprise level businesses. The SimpleWan solution improves overall connectivity. That means improvements to overall traffic speed, VoIP call quality, security and general IT cost and maintenance. Some of the features that SkySwitch Resellers have come to depend on from SimpleWan include: Network Monitoring For all of the reasons stated herein, diagnostic tools are a necessity for resellers of Hosted PBX service. Because resellers of BYOB Hosted PBX have no way to manage or monitor traffic between their network and their customer s. All of SimpleWan s tools are built for advanced diagnostics and provide data for all realtime applications with realtime visual trace routes, MOS scores, latency, jitter, and a matrix style monitoring system. Further, SimpleWan provides detailed bandwidth graphs and historical reporting for last mile carrier troubleshooting. Traffic Management SimpleWan is designed around traffic management and traffic flow required by many of today s realtime services including VoIP, Video, Point of Sale and other real time monitoring systems. These real-time systems are built-in technologies that don t always properly communicate on congested networks. SimpleWan identifies these sessions and makes sure that ALL packets related to these services are sent and received with priority. The SimpleWan system specifically makes these technologies work well on normal or congested DSL & Cable networks without any configuration, right out of the box. The way the Internet works, it s not a matter of if a connection will become congested or have a problem, it s a matter of when. Regardless of if it s a massive upload or maxing out your bandwidth, with SimpleWan, all the real-time traffic will make it to and from it s destination. Fail-Over With today s networks, outages are a real possibility, especially with generic Cable and DSL connections. The SimpleWan solution can provide multi-wan fail-over including full fail-over with a backup connection or a 3G/4G wireless connection. SimpleWan s advanced design can fail-over telephones and computers in as little as eight seconds, restoring full functionality. In conjunction with SimpleWan s proprietary technology, vmpls, the solution can keep an entire wide area network up. HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 10 11

Voice Prioritization VoIP is the standard for business. SimpleWan products have built in, proprietary technology that will keep your phone calls clear and jitter free. Even on low speed connections, SimpleWan can help improve the overall quality of calls. SimpleWan can reserve a set amount of bandwidth when a call is placed, and keep it until the call ends, it s then released to be used by other devices on your network. This prevents one device from hogging bandwidth while simultaneously keeping your calls clear and perfect. These benefits made possible by SimpleWan are available to all SkySwitch resellers, even if they don t currently deploy a Hosted PBX solution from SkySwitch. Conclusion The promise of moving applications, storage and other services to the cloud is that it will lower costs, reduce complexity, improve productivity, all while offering employees the latest features and functionality. The company can maintain focus on the core business tasks, rather than spending time on IT issues. However, adding so many services to the cloud can often degrade the quality of the LAN or expose security issues within their LAN. This is particularly impactful on a real-time service like Hosted PBX. Even a small amount of delay or jitter can severely impact Quality of Service of calls, even when the problem does not originate within the hosted platform. HOSTED PBX WITHOUT THE HEADACHES: QUALITY OF SERVICE PAGE 11 11