White Paper SIP Trunking Abstract This paper provides a brief description of SIP Trunks, how they are deployed and the benefits widely put forward in the marketplace as a reason for adopting them. Analysts have forecast that we are at the bottom of an exponential curve in terms of the uptake in this technology and many suppliers are offering SIP Trunking solutions with similar capabilities. The paper then focuses on TeleWare SIP Trunks which offer a number of unique benefits compared with competitive products benefits designed to enhance your business operations, support decision making and deliver the foundations of a robust business continuity strategy.
What is SIP Trunking? In the context of telecommunications, a trunk is a connection capable of carrying multiple telephone lines/calls at the same time. Traditionally, trunks have been based on expensive ISDN circuits. SIP Trunking provides an alternative way of connecting an organisation s on-site telephone system to the public telephone network and/or connecting a number of sites together to share common access to services, including cloud-based applications. SIP Trunking is widely accepted as replacement for ISDN circuits as a way of achieving cost savings. SIP Trunks are typically delivered over broadband, Ethernet or other leased lines. ISDN Issues Cost and flexibility are the main reasons usually cited for migrating from ISDN to SIP Trunks but, as this paper shows, SIP Trunking has the potential for offering the business valuable benefits far beyond interconnect cost reduction. The flexibility criticism of ISDN circuits is due to their limited scalability which means that organisations using them often pay for unwanted capacity. ISDN2 circuits support 2 voice channels whereas ISDN30 supports 32 channels in total with 30 being available for voice calls. However, a minimum of 8 channels must be ordered (and paid for) on an ISDN30 circuit. This means, for example, an organisation using all 30 channels on one ISDN30 circuit and needing an extra 2 channels to cater for expansion needs to purchase an additional circuit in order to deploy a further 8 channels (2 circuits and 38 channels in total instead of the 1 circuit and 32 channels needed). The high costs of ISDN are partly due to the setup and rental costs and partly to the additional charges levied for capabilities such as resilience, call diversion, setting up of standby sites, out-of-area charges and DDI dual parenting charges. Cost savings of up to 40% can be provided by the channel rental element alone. In addition, many of the capabilities which are viewed by ISDN providers as chargeable options are included as standard within the TeleWare SIP Trunk solution, enhancing its resilience and reliability. SIP Trunking Considerations Digital Subscriber Line (DSL) The use of SIP Trunking over Digital Subscriber Lines is perfectly feasible and often an attractive solution for the SME as well as for extending corporate services to home-based staff. However, there are a number of issues which can adversely affect the speech quality and give rise to variable and unacceptable latency - the delay between speaking and being heard by the other party on the call. For ISDN quality voice, each call requires a bandwidth of 64kbps in each direction. An overhead for the to and from addressing of the voice packets is required by the network protocols which convey the SIP Trunk
over the DSL line. For each call, this adds up to 32kbps; therefore, 93kbps is required for each SIP Trunk channel. Domestic internet connections generally operate over an ADSL line where the upload speed is only a fraction of the download speed. For residential DSL, the speed usually quoted by the service provider assumes a 50:1 contention ratio. This means that up to 50 subscribers could be sharing the available bandwidth during peak periods. Premium quality DSL links can be provided which provide assured bandwidth so contention issues are, effectively, eliminated. This is one of the main reasons that premium DSL links are more expensive that domestic deployments. Using a SIP Trunk over an open internet connection may be acceptable depending on the level of usage on the link, but the open internet provides no guarantee of quality of service. TeleWare can provide a managed ADSL connection which supports up to 14 SIP Trunk Channels with a service level guarantee providing confidence that quality of voice communications will not be an issue. Ethernet Connections and Leased Lines Increasingly, Ethernet point-to-point connections are being offered as an alternative to leased lines. These services are, essentially, pipes of a fixed capacity, such as 10 or 100Mbps, which can be segmented to convey several services such as internet, cloud services and SIP Trunks. As Ethernet connections have a fixed capacity which can be shared amongst the services they convey, they are ideal for use with SIP Trunks, delivering high quality voice. Voice and Data The majority of businesses today are aware of the cost benefits of having a combined voice and data network. SIP Trunking enables this move to a converged network and, for larger enterprise solutions with large numbers of SIP Trunks, this can be highly cost effective. Voice Delivery over SIP Trunking Number Flexibility SIP Trunking provides greater flexibility in the use of numbering schemes. Any geographic and non geographic numbers can be combined in a SIP Trunk and delivered to any office, independent of its physical location. This enables the business to present a local office image in areas where they are not located using local numbering schemes. As SIP Trunk DDIs are held in the cloud, when relocating an office the same numbers can be used irrespective of location. Migration to IP The majority of businesses today are aware of the cost benefits of moving their voice communication to IP. SIP Trunking enables a move to IP services without wholesale replacement of, or upgrade to, existing PBX infrastructure. A mixed architecture solution using both the legacy PBX and IP services delivered over the SIP Trunk will provide a cost efficient means to migrate to IP Telephony rather than having to make a step change or an untimely decision on what IP PBX provider to use.
CLI Presentation In most cases, the use of SIP Trunking means that the CLI (Calling Line Identity) of a calling party is not preserved as the call is passed over the Trunk. This means that the person receiving the call is not aware of the number calling and so cannot use this information in their call handling. This is the same situation as when a PBX forwards calls over the PSTN to an external number. The CLI presented to the final destination is the base number from the outgoing DDI range associated with the PBX. TeleWare SIP Trunking is unique today in enabling the true CLI to be presented to the called party which means that more effective call management can be performed by the receiving party. Announcement-Only Mailbox Often, a company will withhold their CLI for fear that call backs will consume expensive ISDN and PBX capacity. In some situations, the customer may withhold their DDI to prevent call backs to their main outbound number resulting in missed calls, as is often the case with outbound contact centre applications, Hosted Communications can provide an announcement-only mailbox to deal with the return calls. The announcement-only mailbox does not use capacity on the customer s PBX since the mailbox is provided in the cloud. If the SIP Trunk is configured to send an appropriate CLI, when the CLI is called back the caller will get a pre-recorded announcement explaining who the originator of the call was, but will not be able to leave a message. Implementation - Working with PBXs and Using Gateways Hitherto, organisations have typically connected their PBX network to the Public Switched Telephone Network (PSTN) via an ISDN link which is a costly overhead, both in terms of ISDN rental and the PBX hardware required for connection. As PBX vendors are, increasingly, migrating to SIP compliant architectures within their PBX products, a SIP Trunk can often be deployed directly to the PBX using its in-built SIP gateway. Many PBXs in today s marketplace employ SIP as a native feature and SIP Trunks can be implemented without additional costs for hardware and licenses being incurred.* SIP Trunks can be deployed to legacy PBX systems using SIP-to-TDM gateways which preserve investment in the existing legacy PBX systems. TeleWare provided gateways can support many ISDN protocols; therefore, compatibility with existing systems need not be a concern. *Refer to your PBX maintainer for further details.
SIP Trunking for PBX Enhancement SIP Trunking facilitates the enhancement of the feature set already provided by a customer s on-site PBX. A wide range of telephony applications can be hosted remotely on a telephony service provider s secure and resilient platform, such as the TeleWare Hosted Communications platform. Applications can be delivered across the SIP Trunk to any telephony extension, whether IP or connected to the legacy PBX at the customer s site. Where a company is expanding and its on-site PBX is reaching port saturation, SIP Trunking allows IP PBX capabilities from the TeleWare Hosted Communications platform to be delivered to any user extension, both attached to and independent from the customer s on-site PBX. This provides three clear advantages. Firstly, as new users are added, no additional ports are required on the PBX, extending the life of existing hardware. Secondly, users can be provided with access, as required, to a feature-rich set of hosted Unified Communications applications from the TeleWare Intelligent Office product suite on a per user per month basis. This approach enables the business to grow and change to respond to user requirements as needed and without any investment in additional PBX hardware. Thirdly, business applications such as call management can be deployed over the mixed IP hosted and legacy PBX architecture with additional cost benefits being available from the ability to queue calls in the cloud. SIP Trunking for Business Continuity and Resilience TeleWare SIP Trunking includes resiliency and automatic rerouting as standard, greatly enhancing the business continuity of voice connects at no additional cost to your business. In addition to the carrier and network s own resilience, the TeleWare solution employs an additional approach to delivering sustained service through the utilisation of automated number re-routing. This is delivered through enhanced application capabilities, which enable customers to route calls diversely in the unlikely event of IP network disruption. This approach minimises any loss of calls based on direct calls to extensions, as well as calls directed through corporate services, such as IVR. The TeleWare Hosted Communications platform is designed to provide multiple levels of resilience for incoming calls to geographic DDI numbers. The platform collects calls from BT s core C7 network and utilises multiple redundant signalling links to achieve the same high levels of availability that are expected of the PSTN network. The switches in the Hosted Communications data centre are connected into the Hosted Communications IP network using a cluster of high quality SIP gateways, which extend the high availability of the C7 network into the IP core. Calls are passed into Hosted Communications own soft switch architecture ( intelligent Exchange ), which provides a highly fault-tolerant, clustered array of call servers. At this point, the platform provides DDI resilience in 3 stages: Stage 1 The incoming call is passed to the Hosted Communications Session Border Controllers (SBCs), which sends the call into the network via the primary SIP trunk route.
Stage 2 If the primary SIP trunk route s endpoint is not reachable, the SBCs will mark it as temporarily unavailable, and, provided an alternative link exists, will retry the call attempt via any specified secondary route, e.g. a second SIP Trunk if one is in place. Stage 3 If the secondary SIP Trunk route s endpoint is not reachable, the SBCs will mark it as temporarily unavailable. At this point, the Hosted Communications platform will perform an internal lookup of the destination number to identify the alternative DDI that has been configured for this geographic number. Typically, the alternative DDI will be a number that can be reached via the PSTN, such as a mobile. The Hosted platform then makes a call attempt to reach the alternative DDI via either the BT core (PSTN) network, or one of the other carriers to which the platform is interconnected. This provides a further level of resilience, as fallback carriers can be tried for a single call, should any single call attempt fail. After a set period of time, the SBCs will try reverting to Stage 1 in case the primary SIP trunk route is available again, and so on. It should be noted that the second leg call placed over the BT or carrier network, in the event of the invocation of DDI number resilience, will be placed at the cost of the end customer and will be included in their monthly bill. Additional Capabilities Unified Communications The advantages of Unified Communications in terms of increasing staff productivity, facilitating flexible working and collaboration, enhancing customer service and reducing overhead costs have been widely documented. SIP Trunking enables a common set of unified communications applications to be accessible across the whole estate, including satellite offices and home-based staff. No additional on-site hardware is required to deliver these applications, provided the users have access to the SIP Trunk. Applications can be provided as a hosted service on a monthly fee per application per user basis. This allows the business to cost optimise as they are able to pick and choose which applications are required by the different areas of the business. If applications are not used, they can be easily terminated or transferred to other users in the business. Call Recording Recording telephone conversations can improve peace of mind, enhance the understanding of the customer experience and help identify staff training needs. SIP Trunking enables an affordable call recording solution that eliminates the need to purchase expensive on-site recording equipment. Call recording can be delivered from within the cloud providing significant
savings compared to traditional solutions. Recordings can be stored and recalled for reference at any time. Delivery can be on a per user basis or for the entire SIP Trunk. Calls can be recorded on demand by pressing the star (*) key at any point during a telephone conversation and the recording will capture the full telephone call or they can be blanket recorded so that all calls are recorded automatically. In addition to our standard Business Call Recording solution where calls are stored and accessed online using the web interface provided, call recording can be provides with added security features include tamper evident storage, encryption and cryptographic signings of recordings. The secure tamper-evident recording meets the current FSA call recording regulations. Reporting to Enhance Decision Making Management planning and decision making are greatly enhanced by the provision of accurate statistics indicating SIP Trunk usage and, in particular, identifying where missed calls may be limiting the business opportunity. Statistics are provided in standardised report format and can be used to identify where capacity is needed or under-utilised as well indicate where call management capabilities could enhance the performance of the business.
Case Studies The following are based on actual customers using TeleWare SIP Trunking to improve business performance. Customer 1 A large international electronics company wanted to consolidate its call centre operations previously split across four sites and reduce costs whilst improving customer service. All support telephone numbers were ported to the TeleWare hosted platform where calls are answered by an IVR application. A 300 channel SIP Trunk connects the hosted platform to the customer s existing voice network, thereby avoiding PSTN call charges for the onward calls into the customer s call centre. The ability to queue calls on the TeleWare platform frees PBX resource and reduces line costs. Customer 2 A multi-sited company in the automotive sector wanted to maximise its return on investment on a previously deployed VPLS network between sites. The company was able to achieve significant cost savings by stripping out its ISDN lines and running SIP Trunks over the VPLS network from its PBX network and the TeleWare hosted platform. Further cost savings came from using the hosted platform for its PSTN connectivity and taking advantage of more competitive tariffs. Customer 3 This company operates in the television programming and interactive television technology. Viewers call into the company and their calls are forwarded to agents working from home. The company wanted to reduce the cost of operating the service. SIP Trunking was deployed between its on-premise PBX and the TeleWare hosted platform. Outbound calls are forwarded over the Trunks and onto the PSTN via the hosted platform. This reduced the cost of the company s outgoing calls. wptw110301 A TeleWare Group plc company. This document is provided for information only. In line with company policy of continued improvement of products and services, TeleWare reserves the right to alter product specification without notice. TeleWare and Intelligent exchange are registered trademarks. Intelligent Office, Intelligent Number, Intelligent Assistant, Intelligent Connect, Intelligent Mobile and Intelligent Application Builder are trademarks of TeleWare plc. All third party trademarks and registered trademarks are acknowledged. Copyright 2011 TeleWare plc. TeleWare plc TeleWare House, York Road, Thirsk, North Yorkshire, YO7 3BX, UK Registered in England No 4756742 T: +44 (0) 1845 526830 F: +44 (0) 1845 522165 E: enquiry@teleware.com W: www.teleware.com