White paper
January 2010
By Glenn Lowe
Associate Director of Enterprise Product Marketing, Bell
1.0 Introducing SIP Trunking
SIP Trunking is increasingly tipped to be the vehicle that will convey us to the next level of enterprise communications. When Internet Protocol connectivity became a reality shortly before the turn of the century, IP was forecasted to be the carrier of all communication media – voice, video, data and whatever new applications were to come along in the future. SIP Trunking is a key enabler towards the fulfillment of that prediction.
SIP Trunking is getting a lot of attention right now for a couple of reasons. First, the Canadian telecommunications marketplace is in the midst of rolling out productivity-enhancing unified communications (UC) offerings. Second, organizations across the country are in immediate need of substantial cost savings. But due to the scramble for quick savings, early decision-making processes regarding SIP Trunking vendor selection and implementation are based on price rather than the overall value and benefits that unified communications (UC) planning can deliver.
Perhaps because of this increased attention, there is some confusion in the marketplace as to what exactly SIP Trunking does and the benefits that it delivers. This white paper aims to clearly define SIP Trunking technology and to identify and quantify its many benefits. You'll also learn:
- How SIP Trunking can be a robust, secure method of transferring enterprise data
- How SIP Trunking can provide enhanced (UC) services
- Why SIP Trunking is a superior system from the point of view of administration and efficiency
- Clear best practices and technical considerations in moving to SIP Trunking
- The most important considerations in selecting a provider
1.1 SIP Trunking explained
A SIP Trunking call consists of voice over Internet protocol (VoIP) communication that combines transmission of data, voice, video and other applications across Canada, without the need for separate systems for different services, and in some cases, without the need of a corporate wide area network (WAN). Telecommunications manufacturers and providers have adopted the SIP standard as the common protocol in order to facilitate the interconnection of various UC devices.
SIP trunks are primarily used in combination with an IP private branch exchange (IP PBX) to provide public switched telephone network (PSTN) access. They can replace PSTN data connection and primary rate interface (PRI) devices and have the potential to deliver enhanced features and cost savings.
In Canada specifically, interest in SIP technology adoption is growing, thanks to the rapid adoption of IP private branch exchanges (PBXs) in the marketplace. Emerging business grade SIP Trunking networks will take advantage of a service provider's private IP networks to connect SIP-based voice transmissions across in Canada via highly resilient and evolvable network designs.
1.2 How SIP T came to be
Traditional legacy telephony trunking has provided decades of dependable voice service. But because of substantial investments in TDM trunking infrastructure, many VoIP calls to and from the PSTN that could potentially be transferred end-to-end over IP are instead routed over existing TDM networks.
Legacy trunking connects PRIs or analogue interfaces to the PSTN and a physical interface to the central office. Due to this interface architecture, traditional telephony faces physical capacity limitations that do not exist with SIP Trunking. When calls are routed to a non-IP PSTN gateway, the true benefits of IP communication not only go unexploited, but are defeated as analog/digital transcoding causes important signalling information to be lost and creates the potential for degraded quality.
The rise of IP functionality – the functionality that makes unified communications a reality – created a strong demand for a protocol that guarantees reliable delivery of rich IP-based communications. This is why the SIP standard was created: it ultimately connects IP PBXs directly with SIP Trunking service providers, eliminating the need for TDM traffic routing and IP gateways. Unlike TDM-based PSTN signalling protocol, SIP is configured to deliver critical information to more intelligent endpoints than in centralized or TDM-based network components.
Chart: Attributes of SIP Trunking versus TDM telephony
| Attribute | TDM Interfaces | SIP Trunking |
|---|---|---|
| Trunk Interfaces | Physical point to point interfaces to the service provider | As few as 1 interface into the service provider via an IP connection |
| Interface Locations | Directly associated with the location of the PBX | Associated with an optimal location for traffic handoff to the service provider |
| TDM - IP conversion | IP-TDM Gateways to be purchased by the customer | IP-TDM conversion is the responsibility of the service provider |
| Capacity | Purchased per PBX geographic location in increments aligned with T1 capacity | Trunks are purchased per rate centre on an aggregate basis |
| Establishing new PBX locations | New PRI to be physically established to the site from the service provider | Interconnection can potentially be established via software changes (if WAN facilities exist) |
| Moves, adds, changes and deletes | May involve physical interaction | Can be implemented by the service provider via software commands |
Why SIP Trunking?
- No need to connect to a local exchange carrier at every site
- Bandwidth can be provided in any increment; no physical site-by-site connections
- Flexible capacity to meet business requirements
- Rich media transmissions retain relevant UC data
2.0 The benefits: Greater efficiency, increased productivity
SIP Trunking makes communicating more effective and efficient in many different ways. It simplifies administration, saves on infrastructure and makes better use of bandwidth. But while these and other cost cutting benefits are real, the greatest benefit of all lies in increased functionality that translates into greater productivity and efficiency. The many benefits of business-grade SIP Trunking fall into two main categories:
- Architectural benefits
- End user benefits
The architectural benefits of SIP Trunking results in capital and recurring expense savings, not to mention increased flexibility for moves, adds, changes or deletes and the establishment of new PBX sites on corporate networks. Multiple interconnections into the PSTN for a particular rate centre can also be consolidated into a single larger trunk group, resulting in increased capacity utilization, administrative simplification and capital expenditure savings.
The second and more important group of benefits concerns the potential expansion and enhancement of the UC suite of services that go hand in hand with productivity gains and other major efficiencies. In addition to the potential extension of UC capabilities to areas of the organization not connected to headquarters PBXs, properly designed and visionary SIP Trunking for medium to large businesses will also provide enhanced services and attributes to voice calls in the future. As we will see in section 6.0, there are great differences in Canadian SIP Trunking service provider offerings.
2.1 Architectural benefits: Manifest efficiency
While setting up the correct architecture can initially be challenging, once it is up and running SIP Trunking is simpler than TDM in many ways and delivers greater efficiency in terms of maintenance, provisioning and administration. Following are top architectural benefits:
- Simplified infrastructure
PBXs can make use of existing broadband connections to the service provider, limiting the number of connection points needed and thereby simplifying administration of the multiple circuits to the service provider and reducing associated costs. Eliminating PRI and PSTN gateways with a SIP-enabled PSTN interface also serves to:- Consolidate multiple trunk groups into one larger group
- Greatly simplify integration of new PBX or branch sites
- Increase flexibility in the provisioning of trunks
- Simplified administration
With SIP Trunking, the addition of new PBXs or locations to existing VoIP networks does not necessarily require any hardwire changes – the service is IP-controlled and can usually be ramped up very quickly, given that WAN connectivity exists from the service provider or from the corporate offices to the new location. You can also consolidate telephony services to one service provider, simplifying your relationships.
In addition, there are a lesser number of interfaces with the service provider, which reduces the burden of circuit and bandwidth administration on the end user. - Enterprise-grade reliability
Properly designed SIP Trunking services will be as reliable and may in fact perform better than existing trunking services in Canada due to the availability of world class MPLS-based IP VPN services from certain service providers. - Better use of bandwidth
Telephony bandwidth is typically dimensioned to allow for peak usage. That means that some of the bandwidth goes unused at times. A SIP Trunking solution allows you to better account for your bandwidth requirements by having your service provider digitally control the addition of bandwidth or other changes more quickly and effectively.
Because consolidating a greater number of sites into one big trunk results in an increased allowable line-to-trunk ratio due to calling patterns and peaks, SIP Trunking also makes sense from a traditional trunk efficiency perspective. - Linear provisioning
SIP Trunking enables you to secure the precise number of trunks that you need. When increasing capacity on a TDM-based PRI system, for example, blocks of 23 trunks must normally be added, each entailing the need for a new gateway interface port to interact with the PRI link and the PSTN gateway. In a SIP Trunking solution, you can increase your capacity by as little as one trunk at a time by provisioning additional SIP trunks for the edge device via a request to your service provider. - Save on infrastructure
SIP trunks can provide immediate cost savings in most cases by eliminating the need for multiple TDM trunking subscriptions and PSTN gateway interface devices. Ongoing expenditures associated with the ongoing maintenance costs of these devices are also eliminated.
2.2 End user benefits: The core deliverable
While the architectural benefits of SIP Trunking can save money and improve efficiency, end user benefits are the true core deliverable. These include the potential to extend UC capabilities to all areas of the organization, increasing productivity by reducing many forms of business latency and improving access to both colleagues and information. Following are some top end-user benefits:
- Converged communications
The greatest single benefit of SIP Trunking concerns convergence. SIP Trunking acts as a gateway to future carrier and service provider productivity-enhancing applications by integrating voice capabilities with UC tools, including voice, video and web conferencing, unified instant messaging, shared calendars and directories, presence management, file transfer, white boarding and mobile integration. Employees and customers can gain greater access to each other, regardless of time zone and whether they are in the office or working remotely. Employees can also access corporate resources from wherever they are and more easily and simply leverage expertise from colleagues in locations throughout the world. Read more about the various facets and benefits of converged communications in section 3.0. - Improved communication
Another future SIP Trunking end user benefit is flexible calling intelligence – including the ability to associate multiple devices with the same telephone number. An employee can, for example, use an IP phone at their home office, a mobile phone on the road and an IP phone in the corporate office – all of them associated with the same number called. Calls from specific numbers can be flexibly routed to a unified messaging service; trigger an instant message to a mobile device; and also route to another preferred device. This simplifies life for both caller and receiver and reduces business latency: callers reach their targets more quickly and employees don't have to check multiple voicemails. Each user controls this functionality for their own telephone number(s), so it can easily be adapted to meet one's daily business schedule. Adaptable business use of non-business communication devices will also be enabled, ensuring that call billing is associated with the user's main corporate number.
3.0 Get the benefits of UC…WAN or no WAN
Because SIP makes use of the full capabilities of packet-based communications, it provides an evolutionary path for the adoption of UC applications across Canada, whether or not a corporate WAN is in place. The benefits of UC are just beginning to be realized in Canadian organizations, and the potential for incredible increases in efficiency is significant. Following are top UC applications that could soon be provisioned to an organization's every site across Canada through a business-grade SIP Trunking service:
- Intelligent call routing
In the near future, PBX users will be able to assign customized calling attributes based on the incoming calling and terminating numbers. The call can be routed to multiple devices simultaneously based on rules set for the terminating number. Users will be able to engage any of the following, in any priority, based on the originating call: cell phone, office line, voicemail, home line, PDA and/or instant messaging. The end result is reduced interaction latency and an improved ability to service both internal and external clients. - Federated presence
Federated presence allows users on different corporate networks to see each other's availability: in a meeting, in transit, on a call, available, etc. It is a valuable tool that helps colleagues to choose the timing and means of contact. If in a meeting, a colleague might be able to respond to a quick inquiry by instant messenger. Presence reduces latency between corporate interactions in many ways, shortcutting multiple voicemails and emails in many instances. This allows for greater responsiveness, which can resolve issues, break down barriers and increase mutual opportunities within and between corporations, ultimately increasing internal efficiency, competitiveness and customer service. - Remote and IP Remote office
Remote office gives home workers the ability to make a call by clicking to dial on their computer-based client. The home number is then rung, and once answered, the destination number is rung. This allows the call to be billed to a work number. IP remote office extends this service by offering the ability to interface to a SIP-based IP phone with any long distance PSTN calls, again billed to the base office number.
4.0 Best practices and technical considerations for SIP Trunking implementation
While session initiation protocol technology has existed for some time, it has only recently garnered significant interest in the marketplace due to its ability to optimize capital, reduce expense and leverage future UC services. Realizing these capabilities takes planning. In fact, best practices in SIP Trunking involve significantly more network planning than many other technology implementations. Making the move to SIP is not a minor undertaking. An expectation of being able to perform flash cutovers from existing trunking designs would not only be unrealistic, but would also increase the risk of reduced PSTN availability to internal corporate users.
Here is an example of the involved architecture of call flow using SIP Trunking, followed by the steps that must be taken in migrating to SIP Trunking:
Call flow – SIP Trunking toll call
- IP endpoint signals the IP-PBX. This may be SIP or H.323 signalling, or proprietary signalling
- The IP-PBX signals the session border controller (SBC) via SIP signalling
- The SBC signals the SIP Trunking call routing platform via SIP signalling
- The SIP Trunking call routing platform determines that this is a toll call. It also knows the company's long distance provider
- The SIP Trunking call routing platform then routes the call to the appropriate PSTN interface gateway via SIP signalling
- Additionally, the SIP Trunking call routing platform will produce a billing record for this call
- The PSTN interface gateway sends the call to the PSTN via ISDN User Part (ISUP) signalling
4.1 Factors involved in planning a migration to SIP Trunking
There are many architectural factors involved in planning a migration to SIP Trunking: WAN and LAN capacity, firewall settings and compatibilities, number and requirements of sites, different ways of aggregating trunks in various locations, and assessing various benefits associated with specific aggregation strategies. All of these factors play into optimizing a SIP Trunking solution, and every solution will be different, given an organization's specific setup and requirements. Fortunately, there are best practices to rely upon in making the switch. Consider the following:
- Standardization and architectural best practices
The recent upswing in interest regarding SIP Trunking comes in great part thanks to a high level of standardization in the industry, resulting in standards such as the SIPconnect™ Technical Recommendations – a leading set of standards to which a growing number of manufacturers and service providers now comply in order to guarantee interoperability and robustness.
Following the SIPconnect Technical Recommendation is one methodology to achieve SIP Trunking deployment success chiefly because it eliminates issues of interoperability. The SIPconnect standard covers the following areas:- DNS configuration
- Signalling security
- Firewall traversal
- Establishing and maintaining TCP/IP connections traversing network address translation (NAT) gateways
- Authentication and accounting
- PSTN and SIP addressing
- Quality of service (QoS)
- Media handling protocols
- Basic requirements
There are two main components to successfully deploy SIP trunks: a SIP-enabled enterprise edge device enabling PBX-to-service provider interface; and an Internet telephony or SIP Trunking service provider. With these in place, an entrepreneur can create a SIP Trunking solution in a home office. For larger organizations, other factors in creating a successful SIP Trunking solution come into play:- Configuring the LAN for QoS prioritization
- Configuring the WAN (if applicable) for QoS prioritization
- Provisioning aggregation devices with the appropriate architecture to fulfill customers' SIP Trunking requirements
- Configuring firewalls for QoS prioritization
- Devices and aggregation architecture
As an organization evolves over to SIP Trunking, it must account for what, and in which configuration customer premise equipment (CPE) call routing aggregation devices and edge devices will be deployed. Aggregation devices must be able to communicate with all edge devices in the network, regardless of vendor. Aggregation is also a major factor in determining SIP architecture. Deciding on the optimal ratio of aggregation devices to lines – completely distributed, completely centralized, or a hybrid model – depends on a number of factors:- Potential service provider demarcation requirements
- Optimizing LAN/WAN bandwidth
- Testing capabilities
- Services and migration
Certain services may not be compatible with current SIP Trunking – facsimile protocols such as T.38, for example. This highlights the importance of deciding what to migrate over first, and from what site. A planned migration, rather than a flash cutover, carries distinct benefits:- Learn how to administer SIP Trunking in a live situation
- Gain a level of comfort and operational expertise before wholesale switchover
- Voice compression protocols
Different voice compression protocols use different amounts of bandwidth. For example, G.711 packetized uses 81.6 kilobytes per second, while G.729 packetized uses 25.6. A service provider that offers G.729 offers significant benefits to business users, as the interconnection efficiency associated with SIP Trunking is further enhanced. Determining how to hand off to your service provider and at what protocol needs to be taken into account as part of a migration plan. - Points of interconnection and call routing
Optimizing connection points with a service provider depends entirely on your network design. Factors include where the bulk of traffic occurs; and what level of survivability is required. For example, if you had a robust WAN between Calgary and Vancouver, it might make sense to use existing bandwidth to route transmissions internally. In this case, the service provider connection point is of less importance.
If the WAN is less robust (or doesn't exist), you may want to connect at both locations. Planning how to best exploit LAN and WAN bandwidth before handing off to a service provider involves several factors, best worked out in conjunction with a leading service provider:- Traffic analysis
- Long distance usage
- LAN and WAN bandwidth
- Sufficient QoS
5.0 How to select a service provider
Selecting a service provider for SIP Trunking is an important decision for many reasons, chief of which is the functionality that different service providers are able to deliver. Following are primary considerations:
- UC feature evolution
Not all SIP Trunking solutions are created equal. The productivity and customer service benefits of a converged communication network are arguably the greatest benefits of SIP Trunking, and a feature-rich path to future UC services enabled by SIP Trunking is the only way to obtain those benefits. Where some service providers simply provide SIP Trunking to the PSTN, other service providers may offer a feature-rich path to future UC services as part of the SIP Trunking roadmap. - Robustness of design
To what level is redundancy built into the service provider's offering? Is the SIP Trunking service provisioned from two different SIP Trunking nodes that can take over from each other, or is there just one? Is redundancy built into SIP Trunking nodes? Consider also the provider's underlying infrastructure. SIP transmissions run over a service provider's internal multiprotocol label-switching (MPLS) network. What security measures and audits does your service provider enact to optimize the MPLS network availability, and what sort of redundancy exists in the core network? - Interface options
Do you have the choice of connecting to your service provider by IP VPN or by Ethernet? In terms of protocol support, must you conform to one compression protocol, or is the service provider flexible in that regard? - Service and support
How deep are service providers' technical resource capabilities? Do they understand customer premise equipment? Do service representatives have equipment vendor training certification? To what extent are service provider representatives able to help you with evaluation, design, implementation, training and ongoing support? Where does your service provider stand in terms of these capabilities relative to other service providers? - Reporting
There are two types of reporting important to SIP Trunking: utilization and performance. Do service providers furnish reports? If so, are types kinds furnished, only one or none at all? Are they accessible via a common business web portal? Some details on both kinds of reports:- Utilization reports concern call details: the number of calls and where they are going. Without such reports, you might not know that, for example, customers cannot get through during peak times, and that you need to provision more trunks
- Service provider performance reports show just how well you are being served. These typically include mean time to restore service. They are of particular value in the case of a service level objective having been set.
6.0 The Promise of SIP Trunking
Voice networking in Canada is currently dominated by existing TDM infrastructure, which has long filled the needs of Canadian enterprises. However, intelligently evolving your TDM trunking network to SIP Trunking, the open standard for VoIP, ultimately makes it possible to combine the transmission of voice and UC applications across the enterprise or across the country – without the need for separate systems – and potentially without the need for a WAN to interconnect all corporate locations.
SIP Trunking is an enabling technology, bringing the productivity, efficiency and customer service benefits of unified communications to organizations across Canada. UC enablement is the primary benefit, but there are also important benefits in terms of cost savings, reduced administration and infrastructure requirements. Migrating to SIP Trunking is an involved, evolutionary process that requires significant planning. Every customer solution is different, as there are many factors that combine to form an efficient architecture. There are also clear best practices to follow. In selecting a vendor, it is important to keep the key benefit in mind: the ability to deliver enhanced future UC functionality.
SIP Trunking adoption is expected to rise quickly, driving another period of exponential growth in corporate UC adoption. It will enable the next level of efficiency in business through federated UC services, providing better communication and expedited access to information – both of which will deliver tremendous efficiencies and encourage growth.
Talk to Bell
Bell is a leader in the deployment of UC solutions to organizations across Canada. For more information about Bell SIP Trunking services, or for help in evaluating the benefits of SIP Trunking to your organization, have a Bell representative contact you. You can also browse bell.ca/enterprise for information on this and many other solutions.
About the author
Glenn Lowe is Bell's Associate Director of Enterprise Product Marketing. Glenn has 25 years of technical, sales and marketing experience in the Canadian telecommunication industry, with a background in both PSTN-based carrier networks and CPE-based enterprise solutions.