A Kcom GUIDE TO SIP TRUNKING: FEWER CONNECTIONS, MORE CONNECTIVITY. SIP Trunking offers organisations the flexibility to transition from distributed voice network architectures to a more centralised and cost efficient VoIP-based solution. It offers the opportunity to reduce lines, often by 40-60%1, to save money, simplify the network and enhance business continuity. SIP Trunking also enables organisations to move closer to next generation converged communications services. 1 See Appendix
Background The way in which organisations grow and expand depends on all sorts of factors and does not always happen in a structured and logical way. Market opportunities, demographic changes and demand fluctuation do not always follow a predictable cycle. The growth of an organisation is not always organic. Mergers and acquisitions, for example, can mean additional locations have to be factored in, with their own existing systems and IT infrastructures, networks and connections. It is no surprise, therefore, that many organisations have network infrastructures that are far more costly, complex and less efficient than they could be. This is a guide to a better and simpler world; one in which reducing the number of lines through centralisation, and an integrated approach to your organisation s connectivity requirements increases its ability to respond to every call and every opportunity. 21st Century Network The converged network is upon us as BT continues to transition the PSTN network to an IP based system. The modern organisation knows how to harness the benefits of convergence as an empowering factor; driving collaboration, reducing costs, facilitating mobility and home working - treating the network as the extension to the organisation it truly should be. WHAT IS SIP TRUNKING? SIP (Session Initiation Protocol) Trunking connects existing PBX telephone equipment to the public switched telephone network (PSTN) directly over an IP-based data link such as IP-VPN or Ethernet. Since SIP Trunking works with both traditional TDM PBX switches and the new generation of IP-based communications platforms, there is no need to change existing voice infrastructures or telephones. Unlike traditional ISDN or analogue line connections, SIP trunks are not geographically tied to locations and are scalable on a per line basis. This enables organisations with multiple connections from multiple offices to consolidate their lines through SIP Trunks. You can retain your existing numbers and get access to new number ranges from different area codes across the UK. SIP Trunking allows you to integrate and route calls between separate locations in a more flexible and efficient manner. Dynamic failover provides additional business continuity by automatically re-routing calls in the event the normal telephone destination cannot be reached. SIP Trunking also offers customers a future roadmap to additional multimedia services such as presence, IM, web conferencing and video. Effectively, all organisations need SIP Trunking or, to put it another way, all organisations would be well advised to revisit their telephony arrangements from both a technical and contractual perspective as doing so will no doubt unearth savings and efficiencies that can be made. LINE PROVISIONING If you set out to overhaul your voice network arrangements, the start point is to assess precisely how many lines are needed. You need to consider daily traffic and anticipated seasonal or cyclical peaks and troughs in demand, whilst being able to offer the scalability for future growth or tactical responses to unforeseen opportunities or demands. It is also important that you understand the concept of blocking probability. This is effectively the point at which you know your organisation will be turning calls away due to the fact that the physical maximum number of lines is occupied. Over-provisioning means unnecessary overspending, whereas under-provisioning could mean loss of business opportunities or an increase in frustrated customers, so it pays to get the balance right. In the world of telecommunications there are precise formulae to help determine line provisioning requirements, taking into account blocking probabilities. These are based on Erlang and the Erlang tables. It is advisable to understand these conventions before starting any project focused on network connections, line rentals, service provider contract reviews, or related areas. Erlang formulae and tables help clarify expenditure decisions on networks and lines ensuring you enter into negotiations with service providers with a clear understanding of your requirements. An understanding of the landscape according to the Erlang tables enables you to establish control over your network throughout the organisation by adopting best practice and
accepted industry norms for network resource planning. Once these traffic measurement principles are grasped their value in helping validate network and line configuration decisions, and minimising the probability of call blocking cannot be underestimated. Erlang is explained in more detail in Appendix I. HOW SIP TRUNKING WORKS Whilst Erlang calculations will definitively determine the size and scope of the SIP Trunking requirements for an organisation, it is necessary to take one step back to re-evaluate what sort of organisation might benefit most from SIP Trunking in the first place. The key consideration will be costs. High costs of PSTN connectivity for a voice infrastructure invariably arise from a proliferation of connections; lots of connections from lots of locations, usually stemming from a lack of network housekeeping & strategic planning in the past. However, inefficient utilisation of lines and the large overhead to manage multiple contracts is not unique to multi-location organisations and can just as easily apply to a large organisation in a single building or, for example, a campus-style site. There are many examples of organisations unwittingly paying line rental on lines no longer used. Taking a look at a before and after scenario can clarify the way in which SIP Trunking can immediately rectify the situation. Figure 1, before, shows a typical voice connectivity configuration prior to SIP Trunking deployment. Each location connects into the PSTN independently, with a variety of different providers, invoicing procedures, contracts and other administrative as well as technical complexities. In this situation internal calls between locations can be costly, with external calls being subject to expensive voice tariffs due to low volumes per location, because each location is treated, from a network perspective, as a separate entity. You are also likely to experience lengthy provisioning times for new voice lines and long contractual terms which might not necessarily meet the commercial needs of your organisation. Figure 1: Typical voice conectivity configuration before SIP Trunking deployment
A proven methodology for saving network costs. Appendix I offers a quick guide to using Erlang - a measurement unit for voice traffic over a defined period of time (normally one hour). It expresses the volume of total calls to a specified point, which may be a physical location or a PBX system. One hour of continuous voice traffic equates to a call volume of 1 Erlang, which can be made up of one or more calls (i.e. One Erlang could be one call lasting one hour or ten calls each lasting six minutes). USE THE NETWORK TO ACCESS THE FUTURE However, it is more than just the expensive and complex status quo illustrated in Figure 1 that might initiate an investigation into a SIP Trunking solution. The need for diligent and effective network resource planning will also have a significant part to play. Such planning recognises the pivotal role now played by the network in terms of providing competitive advantage and business continuity, in the pursuit of smarter and cheaper ways to drive service excellence. THE MODERN ORGANISATION CANNOT AFFORD CONNECTIVITY DOWNTIME If a location becomes inaccessible or premise equipment fails the loss of incoming calls is simply not an option. Likewise being unable to offer a local presence to geographically spread customers is also important. People feel comfortable dealing with local organisations, and it can help build and improve the relationship. SIP Trunking not only helps to reduce and manage costs, it also helps you to maximise business opportunities. THE BENEFITS OF SIP TRUNKING Cost savings The after scenario, Figure 2, which illustrates an optimised voice connectivity configuration following the deployment of SIP Trunking, Figure 2: Optimised voice conectivity configuration before SIP Trunking deployment
shows a simpler, more efficient scenario. Individual locations are no longer treated as separate entities and are effectively brought together through a VoIP based solution which centralises the connectivity to the PSTN. This brings an immediate business-asusual benefit in terms of highly noticeable cost-savings because calls within your organisation, no matter how diverse the locations may be, are free, as they are considered on-net. Savings also immediately apply to calls with other organisations on the same SIP Trunking Virtual Private Network (VPN); your suppliers and other collaborative partners, for example. Cost savings can also be achieved through improved call tariffs for external calls. Once SIP Trunking is in place you are in a strong position to renegotiate all off-net tariffs with your service providers as they can tender for one consolidated voice minutes contract covering all locations. This would provide significant economies of scale, not previously been possible because individual locations had their own lines with their own individual contracts with perhaps different service providers. Business continuity The virtualised nature of SIP Trunking, as opposed to a static, hardwired, ISDN connection means that calls can be flexibly routed between locations. If you were to suffer an automatic failover in the network all incoming calls can be re-routed to pre-programmed alternative destinations, meaning call will be unaffected causing no detrimental impact to your organisation. The benefits of SIP Trunking are realised because it allows a distributed organisation to be treated as if it were one giant open-plan office. Business continuity is assured if localised communication system issues or connectivity problems arise. For example, if an office building becomes unavailable or employees are unable to travel to their usual work location due to bad weather, calls are automatically transferred to other back-up locations. Flexibility and scalability SIP Trunking also allows for fast provisioning of new lines (assuming that the data access connectivity in place offers sufficient bandwidth), scalable on a per-line basis. This offers organisations with seasonal demand the ability to flex up and down in line with expected demands. It also offers greater flexibility in monitoring and adjusting the number of lines to match the organisation s growth. Travel agents, for example, take on average 50 % of their annual bookings in the first three months of the year. This not uncommon business pattern suggests a complete change in connectivity configuration between two major periods in the year - easily possible using SIP Trunking. SIP Trunking benefits VoIP voice connectivity service Free on-net calls Secure private network access Auto-failover of defined numbers Scalable per line Fast provisioning Numbering freedom Competitive voice tariffs Carrier class SLAs Interoperability with major vendors IP-PBX Online billing reports Next generation technology MIGRATION AND PLANNING PHASES With SIP Trunking providing opportunities to save money, simplify the network and extend capabilities, the next question should be How can I do it? How you go about planning the move from a widely distributed location base, with no integrated number plan (no ability to transfer calls easily between locations or systems), and an abundance of lines and contracts, into a simple, cost-effective, centralised VoIP-based system? Migration to a SIP enabled voice infrastructure, when planned and handled professionally, is relatively straightforward and usually involves no disruption of services. The key project phases you should consider when planning your migration, are as follows: i) Strategic Roadmap - your organisation should formulate a clear vision of its future communications infrastructure requirements, particularly with regard to 21CN features such as presence, IM, web conferencing and video. This should be done in conjunction with other key stakeholders to ensure the communications infrastructure will support the organisation s objectives and challenges in the long-term. ii) Voice Audit prior to making any changes, it is important that you understand your voice requirements and current voice usage (refer to Appendix I); not least so that you can enter into informed discussions with service providers. iii) Network Audit - your data network must be capable of supporting SIP Trunking. SIP Trunking can work with both
traditional TDM (Time Division Multiplier) PBX switches and the new generation of IP-based PBX systems, so there is no need to change existing voice infrastructure or telephones. If a system does not support SIP directly additional media gateways are available. iv) Design and Migration Plan a full run-down of requirements across your organisation, including how it will maximise the new capabilities, what external notifications are required, training needs, testing and implementation arrangements and timeline should be included. v) TCO Cost analysis Full cost implications should be understood for access, line rentals and voice minute tariffs. CHOOSING A SERVICE PROVIDER It is important that a service provider is chosen that can demonstrate expertise and skills in WAN, telecommunications and enterprise IPT networks and equipment. It is also advisable that they have experience in implementing managed SIP Trunking solutions. For example, the service provider should pro-actively monitor availability and quality of connections via MOS (Mean Opinion Scores) as a measure of voice call quality. In addition, the service provider must offer secure private network access as this is the only type of access that can provide quality and security of calls. Public broadband access, whilst cheap, is not sufficiently robust for the needs of most organisations other than very small ones as it is virtually impossible to guarantee or assure quality of call and service reliability. Service Levels need to be Carrier class and cover connectivity, SIP lines and CPE (Customer-Premise Equipment), usually referring to routers, gateways, switches and the telephones themselves. Key decision criteria in choosing your SIP Trunking service provider: Security Resilience Reliability Call Quality Price Customer Experience SUMMARY SIP Trunking opens up a wide range of flexible and scalable opportunities for organisations with multiple distributed sites, and hence a fragmented voice infrastructure. Running independent network connections from offices in different locations is costly and, with the advent of 21CN, is a practice that will no longer have a place in modern organisations. SIP Trunking can reduce costs, simplify the network and enhance business continuity. Migration is unlikely to cause any downtime or disruption - especially if the five planning phases outlined in this paper are considered. During this planning process familiarity with Erlang and the Erlang tables (Appendix I) is recommended to minimise the probability of blocking, and to ensure you have all the information you need to effectively negotiate with service providers. Above all, SIP Trunking is the technology for today and is the future for all switched voice service and end user devices, offering a roadmap to multimedia communications, including presence, IM, web sharing and video. It will serve as a platform in the future to federate to partners, customers and suppliers, offering the benefits of convergence in every sense of the word. ABOUT Kcom Kcom is a leading communications services provider with the largest network reach in the UK and over 100 years experience understanding and addressing the communication needs of our customers. As an independent provider of these services, supported by our BT Wholesale partnership, Kcom is able to offer advanced network capability and related added value services, coupled with a flexible approach to delivering solutions. SIP Trunking is part of our Communications and Collaboration portfolio, which also includes managed LAN/WAN services; professional services including voice audits, health and readiness assessments and advanced design; mobility solutions; and a range of communications and collaboration platforms and applications from market leading vendors such as Cisco, Microsoft and Avaya. Part of the KCOM Group CONTACT US: please visit kcom.com or call 0800 915 5226
APPENDIX I A Kcom GUIDE TO SIP TRUNKING: I. USING ERLANG IN TELECOMMUNICATIONS LINE PROVISIONING Erlang formula are used to calculate the number of lines to a location to provide an acceptable service level without overprovisioning of lines that would be expensive. For organisations with multiple locations SIP Trunking drives up the utilisation rate of the lines that much more efficiently than can be achieved with separate ISDN lines. Centralised voice access involves aggregating individual call volume requirements throughout the organisation and provisioning lines in accordance with the total. Thus, 10 locations of 50 users per location are treated as one virtual group of 500 users. Instead of needing 10 x 10 ISDN lines hardwired to each location (for 1% blocking change for low telephone usage for 50 people location), the equivalent service level can be achieved with 55 SIP Trunking lines through packet switched VoIP, a reduction of 45% in the number of lines required. There are several Erlang formula that differ in complexity, however the most used and simplest is the Erlang B, which calculates the probability of a call being blocked given a number of lines provided to handle a volume of voice traffic. HOW IT WORKS Step 1- to calculate the number of lines required for a specific location the Busy Hour Traffic (BHT) value has to be determined. As calls normally do not come in equally distributed during the day, the Erlang value for peak period needs to be determined, either by observation, call volume analysis or simply an informed estimate based on experience. For example if, during the busiest period of the day 50 users tend to make or receive 3 calls each with an average length of 5 minutes, then the Busy Hour Traffic value for the location concerned would be: 50 x3 x5 = 750 minutes/60 = 12.5 Erlang. more lenient blocking factors of 1 in 50, or even 1 in 20 for less important calls. Step 3- next, a balance needs to be identified between service level (or low blocking change) and cost (or number of lines provided). Especially for smaller locations the cost of providing low blocking factors can become expensive. As an example, a location with 20 people might be supported by a small key system with 4 external lines. Theoretically, as soon as 4 people are on the phone to external parties, no more calls can be received (or made) in that location and any incoming calls would be rejected, receiving an engaged tone. To completely avoid this, and create a non-blocking environment, 20 external lines would need to be installed. This would be a selfevidently wasteful response to the situation, since these expensive additional lines would lie idle most of the time. This is where Erlang comes in, to help find the balance between costs of lines and the service level accepted. Figure (i) Line requirements based on blocking probability Number Number of lines required of Low telephone activity 1 Medium telephone activity 2 people in Blockings Change Blocking Change location 1% 2% 5% 10% 1% 2% 5% 10% 5 3 3 3 2 5 5 4 3 10 4 3 3 3 7 7 6 5 20 6 6 5 4 11 10 9 8 50 10 9 8 7 21 20 18 16 100 16 15 13 11 36 34 31 28 200 26 25 22 20 64 61 56 51 500 55 52 48 43 144 138 129 120 1000 100 96 89 82 273 265 250 233 2000 187 181 170 157 527 514 489 458 Step 2- the organisation then needs to make a decision on acceptable levels of service, or the blockings factor. For example, organisations might decide that no more than 1 call in a 100 is blocked (i.e. receives busy tone and cannot be connected), which would give a blocking factor of 0.01. Other organisations choose Figure (i) shows, for example, that for a small location with 10 people with low telephone activity, 4 external lines are sufficient with a blocking change of less than 1%, but 7 lines would be required to cope with a higher call volume.
APPENDIX I A Kcom GUIDE TO SIP TRUNKING: Figure (ii) Line requirements per user. Number Number of lines per person of Low telephone activity 1 Medium telephone activity 2 people in Blockings Change Blocking Change location 1% 2% 5% 10% 1% 2% 5% 10% 5 0.60 0.60 0.60 0.40 1.00 1.00 0.80 0.60 10 0.40 0.30 0.30 0.30 0.70 0.70 0.60 0.50 20 0.30 0.30 0.25 0.20 0.55 0.50 0.45 0.40 50 0.50 0.18 0.16 0.14 0.42 0.40 0.36 0.32 100 0.16 0.15 0.13 0.11 0.36 0.34 0.31 0.28 200 0.13 0.13 0.11 0.10 0.32 0.31 0.28 0.26 500 0.11 0.10 0.10 0.09 0.29 0.28 0.26 0.24 1000 0.10 0.10 0.09 0.08 0.27 0.27 0.25 0.23 2000 0.09 0.09 0.09 0.08 0.26 0.26 0.24 0.23 Figure (iii) Line requirements by organisation size. Number of lines required Low telephone activity Medium telephone activity 1% 2% 5% 10% 1% 2% 5% 10% Scenario 1: Organisation on 10 branches with 50 people ISDN 100 90 80 70 210 200 180 160 SIPT 55 52 48 43 144 138 129 120 Saving 45% 42% 40% 39% 31% 31% 28% 25% Scenario 2: Organisation with 1 head office with 100 people and 40 branches with 10 people ISDN 176 135 133 131 316 314 271 228 SIPT 55 52 48 43 144 138 129 120 Saving 69% 61% 64% 67% 54% 56% 52% 47% Scenario 3: Organisation with 2 main locations with 50 people and 80 branches with 5 people ISDN 260 258 256 174 442 440 356 272 SIPT 55 52 48 43 144 138 129 120 Saving 79% 80% 81% 75% 67% 69% 64% 56% Scenario 4: Organisation with 1 head office with 500 people and 5 branches with 100 people ISDN 135 127 113 98 324 308 284 260 SIPT 100 96 89 82 273 265 250 233 Saving 26% 24% 21% 16% 16% 14% 12% 10% Scenario 5: Organisation with 3 main locations with 200 people and 70 branches with 20 people ISDN 498 495 416 340 962 883 798 713 SIPT 187 181 170 157 527 514 489 458 Saving 62% 63% 59% 54% 45% 42% 39% 36% Figure (ii) shows that the number of lines per person required to meet the accepted service levels is higher for the smaller locations than for the larger locations, due to the distribution of calls better absorbed in the larger number of lines available. It also clearly shows the impact of the higher service levels on the number of people supported per line. Figure (ii) also highlights the necessity of measuring call volumes as precisely as possible. This may require traffic analyses or asking end-users to log their telephone activities for a defined period. Rule of thumb estimates, allowing, say, 1 line per 5 people on the assumption that it sounds broadly sufficient (a common practice in the area of telecoms), could be completely off the mark, causing high numbers of calls to be missed (blocked) or triggering unnecessary high costs for lines not utilised. The centralisation of voice access through SIP Trunking, aggregating call volumes across an organisation can lead to savings, on average, of up to 45% on the number of lines required. However, as some different scenario calculations in Figure (iii) to the left show, depending on the size and shape of the organisation, savings in the number of lines could go up to as much as 80% for highly distributed organisations. 1 Low telephony activity: making/receiving 1 call lasting average 5 minutes per person in the busy hour period. 2 Medium telephony activity: making/receiving 3 calls lasting average 5 minutes per person in the busy hour period.