Shoretel White Paper on SIP Trunks and Best Practice
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Shoretel White Paper on SIP Trunks and Best Practice

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SIP as just voice, SIP trunking can also serve as the starting point for the entire breadth
of real-time communications possible with the protocol, including instant messaging,
presence applications, whiteboarding or application sharing.

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Shoretel White Paper on SIP Trunks and Best Practice Shoretel White Paper on SIP Trunks and Best Practice Document Transcript

  • WHITE PAPERSIP TrunkingBenefits and Best Practices
  • SIP Trunks - Benefits and Best Practices PAGE 2Table of ContentsAbstract . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 1 What is SIP trunking. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42 The benefits of SIP trunking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52.1 Calculating the investment ROI. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52.2 Bandwidth utilization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62.3 Flexibility to add new lines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82.4 Least Cost Routing (LCR) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92.5 Making IP-to-IP calls when possible . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102.6 SIP trunking – the stepping stone to higher productivity . . . . . . . . . . . . . . . . . . . 113 SIP trunking infrastructure. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123.1 The PBX component . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123.2 The enterprise edge component. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143.3 The service provider component. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173.4 Trunk centralization and virtualization. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184 Interoperability. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184.1 SIP Standards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194.2 Interoperability. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195 Security considerations for SIP trunking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205.1 Threats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205.2 Importance of a stable platform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 215.3 SIP signaling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 215.4 Controlling media. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226 Quality and reliability issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226.1 QoS – Different service provider approaches . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226.2 Prioritization of voice traffic. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236.3 Call admission control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236.4 Poor voice quality can be a client problem, or based on the internal LAN. . . . . . 246.5 MPLS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246.6 Reliability of SIP trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246.7 SIP Trunking may be more reliable . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257 Implementing SIP trunking with ShoreTel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268 Summary. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27The SIP Connect Spec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
  • SIP Trunks - Benefits and Best Practices PAGE 3AbstractBack in the days of wireline telephony, when all phone calls went over the Public SwitchedTelephony Network (PSTN), businesses would purchase “trunks” –dedicated lines or abundle of circuits – from their service provider. Today the new model of “trunking” to IP-enabled enterprises can result in lower telephony costs and a rapid return on investment(ROI) plus the opportunity for enhanced communications within the enterprise as well aswith vendors, customers and partners.A SIP trunk is a service offered by an Internet Telephony Service Provider (ITSP) thatuses SIP to set up communications between an enterprise PBX and the ITSP. A trunkencompasses multiple voice sessions – as many as the enterprise needs. While some seeSIP as just voice, SIP trunking can also serve as the starting point for the entire breadthof real-time communications possible with the protocol, including instant messaging,presence applications, whiteboarding or application sharing.The possibility for a rapid return on investment is a key driver of SIP trunk deployments.SIP trunking delivers the following benefits:•.Eliminates costly BRIs (Basic Rate Interfaces) and PRIs (Primary Rate Interfaces) services.•.Removes the need to invest in additional PSTN gateway capacity as you grow• Reduces capital expenditures: edge devices offer a lower investment path in addingnew lines as they are typically cheaper per line than the corresponding PSTN gateway• Optimizes bandwidth utilization by delivering both data and voice via the sameconnection• Maximizes flexibility in dimensioning and usage of lines as you avoid having to buycapacity in chunks of 23 (T1) or 30 (E1) lines• Provides flexible termination of calls to preferred providers; calls to anywhereworldwide can be made for the cost of a local call.• Enables redundancy with multiple service providers and links.There are three components necessary to successfully deploy SIP trunks:• IP-PBX with a SIP-enabled trunk interface,• Enterprise edge device that supports SIP• Internet telephony or SIP trunking service provider (ITSP).The productivity benefits that come with SIP and SIP trunking are also significant. Byextending the SIP capabilities of the corporate network outside the LAN, satelliteoffices, remote workers and even customers can use VoIP and other forms of real-timecommunications applications to break down barriers of geography to share ideas andincrease productivity.
  • SIP Trunks - Benefits and Best Practices PAGE 4Figure 1 - Two typical SIP trunking solutionsInternetITSPPSTNInternetITSPPSTNSIP-enablingedge deviceSIP-capablefirewall FirewallIP PBXIP PhoneComputersLANIP PBXIP PhoneComputersLANCustomer premises equipment (CPE) based on the SIP protocol – SIP phones, IP-PBXs,etc. – has been around for some time. Now that SIP trunks have gained momentum, it isimportant to ensure that equipment works together. It is for this reason that standardssuch as SIPconnect™ have become so critical. SIPconnect was developed by the SIPForum as a set of best practices for interfacing an enterprise PBX implementation withan ITSP. It attempts to eliminate some of the unknowns and incompatibilities of mixingequipment from different vendors within a single environment.Like any application that opens the private network to the Internet, there are ways tomaximize enterprise security for SIP trunking deployments. One of the most effectivetechniques is to address SIP security the same way data security is addressed – at theenterprise edge. SIP servers and SIP proxy technologies offer maximum control over theflow of SIP traffic, enabling the administrator to ensure correct routing, apply verificationand authentication policies and mitigate Denial-of-Service attacks.Voice quality is not an issue with SIP trunking if proper Quality of Service (QoS) measuresare applied, such as over provisioning of links, and prioritization of voice traffic. Reliabilityis also a moot point because SIP trunks can be more reliable than the traditional PSTNdue to the failover solutions that can be implemented.1. What is SIP trunkingUnlike traditional telephony, where bundles of physical wires were delivered from theservice provider to a business, a SIP trunk allows a company to replace traditional fixedPSTN lines with PSTN connectivity via a SIP trunking service provider on an IP networklike the Internet.
  • SIP Trunks - Benefits and Best Practices PAGE 5To take advantage of SIP trunking, the PBX must have a SIP-enabled trunking interface.As shown in Figure 1, it can be an IP-based PBX communicating to all endpoints over IP,but it may just as well be a traditional TDM PBX. The sole requirement is that an interfacefor SIP trunking connectivity is available.Utilizing the Internet, the ITSP provides connectivity to the PSTN for communication fromboth mobile and landline phones.The IP-PBX on the enterprise networks connects to the ITSP via an enterprise borderelement. The border element could be a SIP-capable firewall or a SIP-aware edgedevice, working alongside an existing enterprise data firewall (all these components aredescribed in depth in section 3).2. The benefits of SIP trunkingA large number of enterprises already use VoIP, however many are only using it forcommunication on the enterprise LAN. In this scenario, VoIP is being used solely as adirect replacement for traditional wire line telephony. For all calls made outside of theenterprise, a PSTN gateway at the enterprise edge is used. These businesses realizea solid return on investment (ROI) just by lowering administrative costs and the costsassociated with calls made within the company.With SIP trunking, the potential for ROI is even greater because SIP trunking takes theidea of VoIP a step further.. The full potential for IP communications can be realized whenthe communication is extended beyond the corporate LAN.The cost effectiveness of a SIP trunk is such that by replacing an installation of a PSTNgateway/PRI with an edge device/SIP trunk, ROI may be achieved in a matter of months.For new installations a SIP capable edge device is most often lower in cost compared tothe corresponding PSTN gateway functionality.2.1 Calculating the investment ROIIt is almost impossible to calculate a “standard” ROI for a SIP trunking investment, asthere are far too many service providers that offer services under widely differing termsand conditions. Additionally, voice calling rates are often bundled with other servicesmaking it more difficult to parse them out separately.This section focuses on the fundamental parameters affecting the costs and theprinciples of how enterprises using an IP-PBX, moving from traditional TDM PRIconnections to SIP trunks, can achieve a rapid return on investment.
  • SIP Trunks - Benefits and Best Practices PAGE 6Figure 2 - Typical bandwidth utilizationTimeTDMInternetMaxMaxTimeBoughtCapacityNumberofcallsNumberofminutes00Peak hourBurstyOne of the immediate ways SIP trunks reduce costs is by eliminating the need topurchase ISDN, BRIs, PRIs or local PSTN gateways or equivalent functionality. Sincethe voice traffic is now routed through the Internet connection to the ITSP, no localconnection to the PSTN is necessary at the enterprise location except for possible afew backup trunks for emergency calling. The gateways needed to connect to the PSTNreside with the ITSP.The devices required at the enterprise edge for SIP trunking are not only typicallycheaper per line than an equivalent PSTN gateway, but they also enable the wholebreadth of SIP-based real-time communication and therefore become a strategic devicein the future of enterprise communication.2.2 Bandwidth utilizationIn many organizations, both telephony and data capacity is underutilized. Telephonyusage patterns in many organizations are characterized by “busy” hours that see manycalls all the way down to almost no calls (e.g. after business hours). Data or Internet traffic,on the other hand, is for the most part characterized by “bursts” of traffic happeningthroughout a business day.
  • SIP Trunks - Benefits and Best Practices PAGE 7TimeTDMInternetCapacity in useCapacity in useMaxMaxTimeslotsNeededCapacityBoughtCapacityWastedCapacityNumberofcallsDataMbit/s00Figure 3 - Comparison of capacity in useFigure 4 - The capacity you need when you need itIf we plot the data using time periods with the highest usage at the left and then indescending order, it becomes evident how much of the total capacity is wasted.In practice, when compared to real-time communications (such as voice), data trafficis usually not as time critical. Combining the two communication types on the sameconnection will give maximum use of capacity. By applying the correct Quality ofService (QoS) settings, critical voice communication can be prioritized over the datacommunication at all times.2Data uses free capacity with the help of QoS settingsBoughtCapacityNeededCapacityfor VoIPFlexibilityto use extracapacityDataMbit/sVoIP-numberofcalls00816251With a SIP trunking solution, the capacity you need when you need it is always available.Instead of dimensioning the telephony capacity for peak usage, it may instead bedimensioned for average usage, allowing the dynamics of QoS to make sure that voicetraffic always gets the capacity it needs. Voice traffic borrows bandwidth from data trafficduring peak usage times.
  • SIP Trunks - Benefits and Best Practices PAGE 8Figure 5 - Costs to expand infrastructure, TDM vs. SIP trunks2.3 Flexiblilti to change line capacityThe cost of adding lines with a SIP trunk connection follows a fairly linear pattern (i.e thecost is directly proportional to the number of lines).MonthlycostUSD4500350025001500500110 150SIP TrunkTDMNumber of lines200When an enterprise using a TDM solution needs to increase its capacity, it generallyhappens in increments of 23 (T1) or 30 (E1) lines. Each additional E1 or T1 span requires:• New PRI subscription.• New PRI connection on the IPBX (PSTN gateway or similar resource).When the capacity of the PSTN gateway resource and/or PRI connection is exceeded, it isnecessary to invest in an additional PSTN gateway resource and/or PRI subscription. Thisis true even if you only need one more line. Going from one E1/T1 to two always requiresadditional hardware and they can generally only be bought in increments of 23/30 lines.Even if you move from an E1/T1 to a higher capacity standard bundle like STM-1, thetermination hardware will need to be replaced. The SIP edge device does not have thesescaling issues.In a SIP trunk solution, the enterprise can increase capacity in increments of a single lineby:• Purchasing additional software licenses for the edge device.• Allocating a greater percentage of the bandwidth for voice. Only if the total bandwidthcapacity is used will the Internet data connection need to be upgraded.
  • SIP Trunks - Benefits and Best Practices PAGE 9Figure 6 - Utilizing multiple service providersThe opposite is also true – if line capacity needs to be reduced due to seasonality in thebusiness, it is just as easy to work with the service provider to turn off SIP trunks therebysaving money.2.4 Least cost routing (LCR)The use of IP makes it possible to cost efficiently use SIP trunks from multiple serviceproviders, depending on optimal availability and the best rates (capitalizing on time zonedifferences, geography etc.). In essence, the enterprise can become its own “MasterService Provider” with subscriptions to service providers in countries where they havethe highest calling volumes. By routing calls to the cheapest service provider based oncountry codes, for example, significant savings can be achieved.ITSPUKITSPSEITSPUSAITSPFRITSPJPPSTN PSTNPSTNPSTN PSTNInternetSIP-capable firewallor SIP capable edgedeviceIP PBXOptional back-upPSTN gatewayIP PhoneComputersLANThese routing decisions can be made by the PBX or by the edge device. The fact that thisability can be built into the edge device means that low functionality PBXs can performrouting functionalities as well. By “outsourcing” this function to the edge device thePBX needs only to send the number as it is, and let the edge device act depending ondestination etc.
  • SIP Trunks - Benefits and Best Practices PAGE 10Figure 7 - Transcoding from IP to TDMITSPITSP SoftswitchPSTNPSTNITSP gateway ITSP gateway Class switchEnterprisePSTNgatewayEnterprisePSTNgatewayIP PBX IP PBXIP Phone IP PhoneIP IPIP TDM TDMUsing multiple service providers provides a higher level of security and reliability:•.Failover to secondary Internet service provider.•.Failover to secondary service provider or back-up PSTN gateway.2.5 Making IP-to-IP calls whenever possibleToday, calls that could be transferred over IP end-to-end are connected through TDMconnections instead. These situations arise when calls are routed to a PSTN gateway.In essence, the true benefits of IP communications are not only unrealized, they aredefeated as quality can suffer due to multiple conversions of voice signals between theanalog and digital domains.ENUM (Electronic Number Mapping System, also known as Telephone NumberMapping) is a standardized address translation technology adopted by the IETF (InternetEngineering Task Force) using DNS (Domain Name Service) to link a phone number toa specific SIP address. This feature is used to automatically look up phone numbers todetermine if they match a known SIP address, allowing the call to be completed over theInternet (instead of transferring it to the PSTN). Since no traffic is placed on the PSTN,ENUM provides an additional means of cost savings for businesses that communicatewith other enterprises also using SIP. If the number is not found in the ENUM databasethe edge device will route the call to the service provider for termination to the PSTN.With the growing installed base of SIP-based IP-PBXs, the critical mass for widespreaddeployment of ENUM will soon be here. We expect that an increasing number of calls willbe transferred directly via SIP over IP between the calling parties instead of going overthe PSTN.
  • SIP Trunks - Benefits and Best Practices PAGE 112.6 SIP trunking – the stepping-stone to higher productivityAlthough it is more difficult to quantify, the productivity gains that can be achieved byutilizing SIP trunking can be significant. Introducing SIP-based real-time communicationhas a tremendous impact on how people work, collaborate and communicate – now andin the future. SIP trunking is an important step in this direction as it is the feature thatmoves communication from the old PSTN connection to the Internet. Once that is donethe field is open for adopting all of the productivity-enhancing features that SIP offers.Rich communicationSIP has become the standard protocol for VoIP. However, it was originally designed toinitiate all types of real-time communications over the Internet, not just voice.These types of real-time communication include:• Presence, to see who is currently online and available.• Instant Messaging (IM), text messaging in real-time.• File transfer.• Application sharing, collaboration on a single document.• Whiteboarding, writing and drawing on a common virtual whiteboard.• Video conferencing.• Machine-to-machine real-time communication.• Distribution of alarm notifications.A broad suite of rich communications options enables users to exchange ideas in thebest possible way for their immediate situation. For instance, remote workers at a WiFi-enabled hotspot may prefer communicating with colleagues via IM, not VoIP.The seamless enterprise – road warriors and home usersOne of the key benefits of rich communication applications is the ability to makebusinesses run seamlessly. Business can be conducted from anywhere in the world –regardless of time zones or locations (such as in the case of remote workers) – so thatcustomers can always reach and interact with your business. In addition, employees canaccess corporate resources from any location that makes them more productive; they canreadily call in expertise from colleagues in other offices or even other countries, or useSIP to provide customers with the best service.This same technology for remote connectivity can be used for all clients including PC-based softphones and IP phones connected to the Internet. This is an advantage of theSIP protocol: the ability to register multiple devices with the same address (i.e. a phonenumber). A user can then, for example, use an IP phone/softphone at their home officeand an IP-only phone in the corporate office, both registered to the same number. Onenumber reaches the employee in multiple locations.
  • SIP Trunks - Benefits and Best Practices PAGE 12Dual-mode handsets supporting voice over cellular and WiFiThe demand for mobile phones equipped with both cellular and WiFi capabilities is verystrong – all smartphones sold today offer this capability. The potential cost savings for aperson who frequently travels overseas, and is able to transform expensive cellular callsinto near zero cost VoIP calls when connected to the Internet, is significant.Mobility solutions that seamlessly switch between WiFi and cellular (3G) connections withno interruption in the call are available today and extend a rich set of capabilities to themobile workforce.3 SIP trunking infrastructureThis section will describe in detail the three components needed to set up a SIP trunkingsolution:• PBX.• Edge device that can handle the traversal of SIP traffic .• SIP trunk from an ITSP.3.1 The PBX componentThis section will provide an overview of the different types of PBXs available on themarket.The traditional PBXA PBX (Private Branch eXchange or Private Business eXchange) is a telephony exchangeserving an enterprise or large branch location. It performs the basic function of routingcalls to their destination as well as providing a large number of value-added features: calltransfer, hold music, redirect when busy or no answer, etc. The traditional TDM PBX wasconnected to a dedicated premise network that only carried voice traffic.The line-side IP-enabled PBXThe LAN for data traffic was a later addition to the enterprise and was deployed as aseparate and parallel premise network. For many years these two networks coexisted,serving separate but related communications functions.The first IP-based PBXs, or IP-PBXs, focused on making the line side of the PBX, (i.e. theside connecting to the telephones) run over IP. The first and very obvious advantage ofdoing this was that the two premise networks now could be converged into one commonnetwork – the LAN. By using of IP phones, these could be connected to the samephysical networks as computers and servers. Having made this change to a commonpremise infrastructure, it also became possible to introduce PC-based soft phonesinstead of traditional telephone sets.
  • SIP Trunks - Benefits and Best Practices PAGE 13Some argue that voice and data traffic should not be mixed on the same LAN or at leastshould be run on separate virtual LANs (VLANs). The reason for this position is thatvoice traffic, due to its real-time nature, is sensitive to delays or lack of bandwidth in theinfrastructure, which may result in poor voice quality. However, this issue is readily solvedand should not stand in the way of realizing the benefits of converged communicationas described in the previous section. The bandwidth available on most enterprise LANs,100Mbit/s or 1Gbit/s, is more than sufficient for most typical enterprise applications. Byusing appropriate QoS techniques, enterprises can easily ensure that the voice trafficgets the appropriate priority to ensure voice quality.The IP-PBXAs mentioned above, IP telephones connected through the corporate LAN have beenaround for quite a few years. However, whenever calls needed to flow outside thecorporate LAN they had to be routed to a local PSTN gateway (or through a PSTNgateway function within the PBX) and converted to traditional TDM-based telephony.This often requires proprietary equipment, which can be expensive, especially as TDMcapacity is added.In a world where more and more endpoints are IP based, there is a risk of deterioratingsound quality due to repeated transcoding between IP and TDM as shown in figure 7 insection 2.5.The next natural step in the evolution of IP based communications, is to use IP for theinterface to the world outside the corporate LAN. This is done by IP-enabling the trunkinterface on the PBX as well, thereby completing its full transformation into an end-to-end IP-PBX. In practice, this happens in one of two ways. For earlier TDM or IP-PBXsthis can be achieved by placing an IP front-end on the trunk interface creating what isusually referred to as a hybrid IP-PBX. This PBX contains both legacy TDM and IP-enabledparts. Newer IP-PBXs, or systems that are designed from scratch, are usually built withIP technology from the ground up, without the legacy TDM part. For such systems anyconnection to the PSTN requires a distinct PSTN gateway resource.There are a number of protocols available that could be used to IP-enable the trunkinterface, including MGCP, H.323 and SIP. The use of an IP-based trunk interface providesall the benefits described in the previous chapter and addresses the issues of soundquality and cost.Benefits of IP-based PBXs over legacy systemsIn the following section further advantages of IP-based PBXs, in addition to the benefitsof SIP trunking, are highlighted.Connect multi-vendor end pointsThere is a trend in the PBX market to allow equipment from different vendors to coexistwithin the same PBX system. This will allow the enterprise to preserve investments made
  • SIP Trunks - Benefits and Best Practices PAGE 14in phone endpoints even if the central PBX equipment is replaced. This allows the user toselect phones, media servers and switches from their preferred vendor. PBX vendors thatchoose to allow this believe that the customer will be more likely to swap to their systemif they can keep their existing phones. Some vendors, however, continue to lock theircustomers in to their own end equipment by making various proprietary extensions tothe system.User managementOne of the most obvious advantages of an IP-based PBX system is increasedmanageability:• By using the existing data network the need for separate wirings for a telephony systemis eliminated.• The phone becomes a kind of computer that allows the administrator to easily makeupgrades and force policies to each phone from a central management system.• The ID and configuration of the phone will follow the phone, regardless of where it isconnected to the network.• Users may log in to the phone when they arrive at a new desk; user profiles andinformation will automatically be loaded into the phone allowing greater flexibility.Integration with other IP-based applicationsThe SIP IP-PBX serves as the primary registrar of SIP users and utilizes this informationfor routing purposes. But the fact that the PBX is now IP-based also means that it can beintegrated with other communications applications running on servers on the LAN. Oneof the best examples of this is converged communications soft clients that can integratevoice capabilities with applications such as video conferencing, presence, instantmessaging, file transfer, white boarding, etc. Through such integration the PBX becomespart of a greater converged communication system that enables the enterprise to benefitfrom productivity enhancing communications applications.3.2 The enterprise edge componentThe enterprise edge component can either be an existing firewall with comprehensivesupport for SIP or an edge device connected to the firewall, handling the traversal of theSIP traffic.Firewall/NAT traversalWhen moving to VoIP, the telephones are IP enabled and connected to the data network.It is imperative to safeguard the system from attacks and other unwanted access. A datafirewall protects the network by rejecting attacks and illegal data packets, allowing onlyapproved traffic. On a local area network, where several PCs or other equipment areconnected, it is common to have private IP addresses on the LAN and a single common
  • SIP Trunks - Benefits and Best Practices PAGE 15public IP address to the Internet. Data that flows between the private and publicnetworks has to pass through a NAT (Network Address Translation) function, which isusually integrated into the firewall.Firewalls and NAT routers are designed for data traffic that is initiated from the insideof the private network. Because malicious attacks on the network frequently originatefrom outside of the private network, firewalls and NAT routers protect the enterpriseby blocking this kind of traffic. Often inbound SIP traffic is not recognized by traditionalenterprise firewalls or NAT routers and can be rejected as unwanted traffic.The biggest hurdle for IT managers looking to SIP-enable their network is architectingthe system to handle the traversal of SIP traffic across the firewall. The majority of currentfirewalls and NAT-routers are still not designed to handle full end-to-end communication,and inbound voice traffic will not reach users on the LANs unless the enterprise firewallhas specific SIP support. SIP traversal of firewalls and NATs is becoming a commodity inthe sense that most vendors advertise support for the protocol. However, the basic SIPsupport offered by most of these vendors does not have the richness of features to fulfillthe needs of a complex enterprise environment. It is critical that IT managers evaluatetheir current firewall solution to ensure there is proper SIP support when new firewallsand NAT routers are installed.One problem is that the media streams (e.g. voice) are transferred over dynamicallyassigned UDP ports that are generally closed. The firewall must be able to dynamicallyopen and close ports based on the negotiation that happens via SIP signaling when a callis set up. Another problem is that the endpoints inside the firewall cannot be reachedby IP addresses since these are generally private and local to the LAN. Communicationsimply cannot happen, unless there is specific SIP support in the firewall.Several methods have been suggested to resolve the issue of reaching endpoints on theLAN. One such method solves the problem where it occurs – within the firewall itself.Firewalls that have a SIP server, with SIP proxy, SIP registrar and possible B2BUA (Backto Back User Agent), which dynamically control the firewall have been available for manyyears. This solution provides optimal flexibility as SIP signaling can be rewritten andprocessed in a very flexible way ensuring correct routing and interoperability with othersystems built to RFC 3261 and related standards.Several firewall vendors develop models with SIP ALG (Application Layer Gateway). ALGsusually work at a lower level than a proxy, adjusting the data packets “on the fly.” Majorvendors have developed firewalls with ALGs that also handle incoming calls to multipleusers, while more simple implementations may only support a single user on the LAN.One limitation of the ALG architecture is that it cannot handle secure SIP signalingvia TLS (Transport Layer Security). This architecture also lacks the ability to rewrite SIPsignaling in several ITSP scenarios.
  • SIP Trunks - Benefits and Best Practices PAGE 16Mediation between PBX and service provider equipmentMost basic call scenarios in a SIP trunking solution, using equipment from differentvendors, work well. However, when more advanced features such as call transfer are used,problems occur when vendors do not strictly adhere to the standard. In addition, SIP isa flexible standard that leaves some room for vendor interpretations. This means that, attimes, two entities can have difficulties talking to each other even though none of themdirectly violate the standard.To make the situation even more complex, some ITSPs and PBX vendors only implementparts of the standard. Or, they add vendor-specific extensions to the standard.While performing traversal and security, these SIP-capable edge devices can alsomediate between the PBX and service provider, offering an important function. They canprocess the SIP signaling and media in a way that is understood and expected by boththe ITSP and the PBX.Security from the edge deviceSIP-enabling edge devices can also add a layer of security to enterprise communications,specifically in securing SIP media. Most security administrators will have serious concernsconnecting a PBX system directly to the public Internet without any SIP-aware firewall infront of it. Like any server on the LAN, it needs to be protected by a firewall. A PBX is notbuilt to withstand or recover from denial-of-service attacks and, in most cases, does nothave filtering capabilities available to reduce traffic (requiring processing power to onlythe appropriate traffic). The enterprise edge device can secure the SIP media as well asdata traffic.The edge device can also protect the network from eavesdropping. Solutions forencryption of media and signaling using IETF proposed standards are recommended.These solutions include TLS (similar to SSL used for https) for signaling and SRTP (SecureReal Time Protocol) for media. Both are recommended in the SIPconnect initiative.Branch office interconnectWhen the PBX is IP-based, a whole host of new possibilities open up since communicationbetween the PBX and other devices (including phones) are using a protocol (SIP) thatworks just as well over the Internet as on the corporate LAN. This means it is nowpossible to connect with other offices within the same organization or with partners andcustomers via IP – without the need to traverse the PSTN network and without the needfor dedicated circuits. This actually enables an entire, multi-site enterprise to use onecentrally located IP-PBX instead of installing separate PBXs at each site.When doing branch office interconnect of SIP-based systems, the same problems oftraversing the corporate firewalls and NATs as with SIP trunking itself will occur. A SIP-capable enterprise edge device will solve this problem as for SIP trunking. Some peopleeven refer to such an inter-office connection within an enterprise as a SIP trunk.
  • SIP Trunks - Benefits and Best Practices PAGE 173.3 The service provider componentA traditional voice telephony service provider typically offers one or more T1/E1trunks to the enterprise for fulfilling its needs for voice communication outside itsown premises. The service provider is then connected to what is sometimes referredto as “the world’s biggest machine”: the worldwide PSTN. Connectivity among thenetworks of the different service providers that constitute this “machine” is achieved bybilateral interconnect agreements between the various service providers. There are alsowholesale service providers that aggregate the traffic from several local service providersand make the interconnect agreements for all of them collectively.The SIP trunk offering is just another way of connecting the enterprise subscriber to thenetwork. The interconnect and wholesale aspects remain the same. In a SIP trunk, thetraditional T1/E1 interface (“trunk”) is replaced by a SIP-based connection that runs overthe Internet connection to the enterprise. Nowadays, most enterprises already have sucha connection for their data traffic. As SIP trunks are software and IP based, they are easierto manage remotely and therefore cheaper for the service provider to maintain. It alsotypically does not require the service provider to deliver and take responsibility for anyadditional customer premise-based equipment. That too adds to the simplicity and costeffectiveness of SIP trunks as a means of delivering PSTN connectivity.Different types of SIP trunking service providersLong gone are the days when there was only one carrier available to offer telephonyservices. These “old” incumbents are, however, still there and they do offer SIP trunkingservices. These service providers typically have their own facilities all the way down tothe subscriber, which means that they have greater control over the quality of the servicedelivered. However, as discussed in the quality-of-service section that is by no means theonly way to ensure that voice quality is maintained in a VoIP network.Among the newer entrants to the voice market, offering SIP trunking and other VoIPservices, are facilities-based and facilities-less providers. Generally there are only a fewmajor companies that have their own network infrastructure while others are resellingtraffic that will travel on another party’s (be it a “new” IP wholesaler or an incumbent)network. The number of such VoIP resellers is increasing rapidly because in the IPenvironment delivering such a service is relatively simple, at least in comparison to theold TDM world. With this arrangement the customer gets the best of all worlds: thefacilities-based operator can focus on operating a high-volume large network in the mostefficient way while the reseller can focus on customer support, billing simplicity and othercustomer-related features of the service.The move to IP also enables service providers to create bundled offers. There are severalcases where an Internet Service Provider (ISP) adds a telephony service to its offering.Such an Internet Telephony Service Provider (ITSP) can create attractive bundles of dataand voice capacity making use of the bandwidth utilization benefits.
  • SIP Trunks - Benefits and Best Practices PAGE 18PSTN connectivityA SIP trunking service provider aggregates the traffic from many enterprise customers.The traffic passed to the PSTN is of much larger volume than the traffic from anyindividual enterprise. This means that the SIP trunking service provider can acquirethe call minutes from the PSTN service providers at a lower rate than the individualenterprise. The network charge for the IP part of the call is typically not traffic-dependentso there are significant gains to be made here.Local breakoutThe use of the IP networks for certain sections of the call means that a service providerwith several points of presence around the world, or that has agreements with otherservice providers to exchange traffic with, can allow the call to stay on the IP network foras long as possible. The call is transferred to the PSTN at the point of presence closest tothe destination of the call. This process, sometimes referred to as “local breakout,” allowsthe service provider to make maximum use of local PSTN call rates rather than payinginternational or long distance charges. This contributes to making SIP trunking a verycost-effective solution for the enterprise as well as for the SIP trunking service provider.3.4 Trunk centralization and virtualizationState of the art SIP trunking deployments commonly centralize trunking to the enterprise.This means that a geographically distributed enterprise will terminate the majority ofits trunking, via SIP, at a central or headquarter location and calls are then distributedto other locations via the corporately owned and managed WAN. This eliminates theneed for maintaining services with multiple service providers in different geographiesand allows the enterprise to cost optimize its trunking by negotiating with a single ITSP,thereby leveraging economies of scale.By virtualizing the SIP edge device, even further cost savings are possible. The edgedevice would be deployed as a virtual machine and can leverage existing investments inserver host capacities.4 InteroperabilityOpen standards are the key to success of voice over IP adoption. Back in the mid-1990sboth email and Web browsing became ubiquitous practically overnight, driving themajority of people in the industrialized world to connect to the Internet. Many differentorganizations, companies, universities and individuals contributed to the exponentialgrowth. With all these different stakeholders, the success we see in this area would nothave been possible without the adoption of open standard protocols like SMTP andHTTP – both developed by the IETF.
  • SIP Trunks - Benefits and Best Practices PAGE 19As mentioned before, the open standard for VoIP is the IETF standard, SIP. SIP-basedreal-time communication over the Internet will be one of the drivers for another wave ofInternet growth.4.1 SIP standardsSIP stands for Session Initiation Protocol; the name describes well what it does. It isused for setting up sessions between endpoints. Endpoints are often end-user devicesor servers. SIP differs from the signaling protocol of the PSTN domain in that it allowsfor locating much more intelligence in the endpoints rather then in centralized networkelements.SIP is specified in a growing number of IETF RFCs. In order to aid the reader navigatingthrough the various RFCs a “hitchhiker’s guide” to SIP has been created by the IETF. Thatguide, and an extensive list of references, can be found at the end of this white paper.Different groups with varied interests have taken part in adapting these standards. Someare PSTN operators who (in some cases) try to redesign the PSTN world on top of SIP.Mobile operators, 3GPP, IMS, as well as companies focused on data communication or IT,push for support of features like IM, presence, file sharing, video etc.As the SIP standard is comprised of a large number of specifications, most vendors donot implement all of them. SIPconnect is an example of how a specific subset of thesespecifications can be used for defining a limited feature set (in this case, SIP trunking).4.2 InteroperabilityEven though the SIP standard is written with interoperability in mind, integrating SIPequipment from different vendors always takes time because, all too frequently, thereare minor inconsistencies with regard to how the different vendors interpret the SIPspecifications.With regard to SIP trunking, different operators will utilize equipment from many gatewayvendors who have varied requirements when it comes to the authentication of the SIPtrunk user. If a company is looking to use SIP trunks from more than one vendor, (e.g.in order to implement least-cost routing) they would normally have to deal with thecomplexities of interoperability among several SIP trunks that each behave in differentways.As mentioned above, enterprise edge devices can mitigate these issues by addressingthe complexities of interoperability. The device handles these details and the differentways to handle authorization for the SIP trunks. From the inside, the edge device willappear as one SIP trunk, even though it will then distribute traffic to several SIP trunksfrom different vendors on the outside. As the customer device that is located closest tothe operator, an edge device is well placed to handle this type of operation.
  • SIP Trunks - Benefits and Best Practices PAGE 20Another interoperability problem common with SIP trunking is when one endpoint islocated behind a SIP-unaware NAT box (home user, hotel, etc.). When the edge deviceis the first point of contact for such an endpoint, remote connectivity technology canenable such users to participate in both outbound and inbound calls even though theyare behind a SIP-unaware NAT.Call transfer represents another interoperability problem. Some operators do notsupport this feature, and some SIP user agents do not support it either. Additionally, auser who has a phone that can support call transfer cannot detect if the phone on theother end does so as well. If a call transfer attempt is made and fails, the call is oftendropped.Edge devices can detect when a call is being made to or from an endpoint that does notsupport call transfer. If someone still attempts to transfer a call to or from that endpoint,the device can perform the transfer itself, in lieu of the endpoint that is not able to. Thecall will be transferred, and the edge device makes sure that the media is sent to anotherdestination. By using B2BUA in the device the party that does not support call transferwill still think that they called the intended person.5 Security considerations for SIP trunking5.1 ThreatsConnecting a device to the Internet exposes the entire network to many types of threats.One example is a brute force attack where the intruder tries to log into a service using auser/password database trying a huge number of username and password combinationsuntil the intruder finally succeeds in finding the right one. Once access has been grantedthe intruder may be able to launch other types of attacks based on known vulnerabilitiesto the service in question and in this way get access to other services or data.Another example of a threat would be Denial of Service (DoS) attack where the attackeruses many different hosts or bots to send a large number of packets to make the hostdrown or crash due to the vast amount of traffic.The above are two examples of traditional data communication attacks. These and manyothers can easily be transformed into attacks on VoIP equipment. The VoIP SecurityAlliance or VOIP-SA has categorized possible attacks and threats on a VoIP system andmade this information publicly available. This document is a resource for understandingwhat threats needs to be taken into account when it comes to securing VoIP in SIPtrunking scenarios.
  • SIP Trunks - Benefits and Best Practices PAGE 215.2 Importance of a stable platformFirewall vendors have developed significant expertise in securing data communication.They know how to design stable systems that are locked down to only admit services thathave been configured to pass. Firewalls inspect and log traffic and, if intelligent enough,they can even block suspected attacks including traffic from known bad sources.Firewalls alone cannot prevent Denial of Service (DoS) attacks, but they can be built towithstand attacks, making them harder to occur. Firewalls can also lay the foundation fora swift recovery. More importantly, they can be built to protect the enterprise LAN frombeing reached by the DoS attack.5.3 SIP signalingFirewalls with a SIP server and full SIP proxy play a critical role in maintaining enterprisesecurity, and securing SIP trunks. They can rewrite SIP signaling and process in a veryflexible way, ensuring correct routing and interoperability with other systems built to RFC3261 and related standards.One important part of the SIP proxy is the SIP parser. The SIP parser verifies that theSIP message is valid and that it may be forwarded to the local LAN. Malformed SIPmessages are discarded. The SIP parser must be robust enough to withstand any typesof malformed SIP messages without crashing. Also, to mitigate DoS attacks, the parsershould be able to process a very large number of packets.The SIP proxy should include support for the optional loop detection mechanismdefined in the SIP specification. This mechanism discerns whether a SIP message islooping (sending the SIP message to itself) and, if so, aborts this behavior. This detectionmechanism also protects against DoS attacks where a SIP message is constructed tocreate loops and thus keep the SIP proxy too busy to engage in useful processing.SIP signaling consists of messages in ASCII text (plain text), and are therefore easyto read and manipulate. It is strongly recommended to encrypt and authenticate SIPsignaling. This is normally achieved by supporting TLS or MTLS. MTLS is the most securemethod as both server and client mutually authenticate each other using CA-signedcertificates or certificate chains.In order to provide greater and more flexible protection mechanisms, filters are usefulfeatures. A typical filter would include the following:• SIP methods allowed or prohibited on the network.• Authentication enabled or disabled per network and SIP method.• SIP messages filtered based on content type.• Incoming callers can be restricted to a white list; this list can be individually enabled/disabled per user.•.From/to header may be used to allow or disallow processing.
  • SIP Trunks - Benefits and Best Practices PAGE 225.4 Controlling mediaSIP proxy technology is an excellent way to add a level of control to the flow of SIP media.This control offers tremendous advantages with regard to security.The main purpose of SIP is to set up a media session between clients. Media is handledby other protocols (often RTP). For media to traverse the enterprise edge, the SIP proxymust dynamically open the media ports for media to flow during the duration of the call.As soon as the call is completed the media ports are closed. This behavior is much moresecure than solutions with non–SIP-aware firewalls/border elements where a media portrange constantly needs to be open. In general the SIP proxy approach is more securethan the IETF specified STUN/Turn/ICE methods, which requires that ports are left openfrom the inside of the firewall to allow media port negotiation to succeed.In addition to the dynamic opening and closing of media ports, the edge device shouldonly accept incoming media from the endpoint that receives media from the edgedevice. This protects against hackers trying to inject media from other endpoints ordevices.To protect media from being overheard by unauthorized persons, media encryptioncomes into play. The industry seems to have chosen SRTP using descriptors for keyexchange as the de facto standard for media encryption. Using SRTP to encrypt mediatraversing the Internet effectively stops eavesdropping. The integrity of the call is muchstronger than ever possible on PSTN.6 Quality and reliability issuesOne of the main concerns about VoIP and SIP trunking is with regard to Quality of Serviceand reliability. Will voice quality be good enough? Will the telephony service be availablewhen I need it? The answer to both questions is a definitive yes. In fact, many people whouse traditional PBXs are using VoIP without knowing it, as many service providers use IP intheir backbone networks.Clearly, IP is not the issue. How the network is managed and planned is what makes thedifference.6.1 QoS – Different service provider approachesThe bottleneck on the Internet is often the last mile connection to the enterprisepremises. There are two methods used by service providers to deliver adequate Qualityof Service. In theory only the service provider controlling the link the entire way wouldbe able to guarantee an adequate level of Quality of Service. However, in practice,the service provider relying on the over-provisioning of links will also be able to offerexcellent quality.
  • SIP Trunks - Benefits and Best Practices PAGE 23Service provider controlling the connection all the wayIn this case, the service provider owns the connection and can control the equipment allthe way from the enterprise to their SIP trunking PSTN termination point. This makes itpossible to prioritize the voice traffic over data and also to give different Service LevelAgreements (SLA) for different customers.Over-provisioning of linksHere, the SIP trunking service provider facilitates the connection all the way to thesubscriber. Any Internet connection is possible as long as there is enough bandwidth.Good voice quality is achieved by over-provisioning of the link so that the last mile neverbecomes a problem.6.2 Prioritization of voice trafficTo maximize the utilization of a given capacity, both data and voice should be deliveredin the same connection. However, this makes prioritization of the voice traffic necessary.Prioritization, which can take place in the firewall or edge device, can be based on:• Services (protocol and port).• Packet size.• SIP traffic.• IP-address and segments.This prioritization should be possible for both outbound and inbound traffic. It shouldalso be dynamic so that bandwidth dedicated for voice can automatically be used fordata when it is available.The setting of Type of Service (TOS) and/or DiffServ bits on packet level will makeit possible for routers on the Internet to make prioritizations. There is no guarantee,however, that all pieces of equipment on the Internet are using these settings forprioritization. In this case it will, of course, help if the service provider controls thecommunication all the way out to the customer premises.6.3 Call admission controlCall admission control, also implemented in the edge device, ensures it is not possible toinitiate more calls than what should fit into the link. The administrator defines the amountof bandwidth that is dedicated for voice and the bandwidth per call based on the codecused for voice. The edge device then keeps track of all calls and when the dedicatedbandwidth is used no additional calls can be made or received. The response from theedge device in this case will be “service unavailable.” It is important to reserve call slot(s)for emergency calls.
  • SIP Trunks - Benefits and Best Practices PAGE 246.4 Poor voice quality can be a endpoint problem, or based on theinternal LANAnother overlooked factor is QoS on the internal LAN. If the LAN is the bottleneck, thevoice quality will be poor no matter how good the quality of the Internet connection maybe. It is highly recommended that customers work with their vendors and complete athorough network assessment prior to deploying their IP Telephony system.6.5 MPLSMany operators offer MPLS as a means of delivering QoS in a VoIP service. The MPLSnetwork is a service provider-managed VPN. However, it is as easy to achieve goodQuality of Service in an open standards-based SIP trunking connection as with MPLS.One of the most important factors is whether the service provider controls the links allthe way from the enterprise to the PSTN termination or not, not which protocol is used.Also, SIP trunks are sometimes delivered over an MPLS connection for voice only. Thismeans there is no support for global SIP connectivity over the Internet and the solutioncan never be more than just a one-to-one replacement of the traditional TDM lines.6.6 Reliability of SIP trunksAnother argument commonly heard is that a SIP trunking connection is not as reliable asthe traditional TDM. It is true that Internet connections are more dependent on electricalpower, and TDM lines may have a slightly better average uptime in many parts of theworld. However, many enterprise telephony systems also rely on electrical power, so apolicy with uninterruptible power supply (UPS) that corresponds to the desired uptimeis a must. Furthermore a TDM line, when down, is truly dead. With SIP trunks alternativebackup solutions are available.The migration to SIP trunks will not happen gradually, so the enterprise might optionallychoose to keep some traditional TDM/PSTN gateway capacity for backup and/oremergency calling purposes.With the right choice of redundancy features and service provider, SIP trunking may evenoffer higher reliability than many TDM-based networks.
  • SIP Trunks - Benefits and Best Practices PAGE 25Figure 8 - Comparison of TDM solution and a SIP trunking solutionITSPThe SIP trunking wayThe TDM waySIP and data trunk from each officeCentralized PRI trunksHeavy loadSingle point of failureVPN to HQ PSTN gateways Exactly the capacity you need when you need itHQHQ6.7 SIP Trunking may be more reliableDue to the inflexibility in the TDM in terms of number of lines, it is tempting to have acommon PRI pool of lines at the headquarters also serving the branch offices with PSTNconnectivity.Many IP-PBX installations look like the left side of the figure above.This will provide a single point of failure combined with an unnecessary high load at theheadquarters. The SIP trunking scenario on the right offers higher reliability (here, withthe different sites independently connected to the SIP trunking provider).In many cases a SIP trunking connection may be more reliable than the traditional TDMin itself. It offers more backup alternatives including the ones described in the followingsections.Failover to secondary SIP trunking providerWith SIP trunking it is possible to utilize multiple service providers for PSTN termination.The edge device handling the SIP trunking connection should be able to automaticallyfailover to a secondary (or tertiary, and so on) SIP trunking provider if the connection tothe primary service provider fails.
  • SIP Trunks - Benefits and Best Practices PAGE 26In addition, to make the switch triggered by a failed call, the device should be able tomonitor the primary service provider by periodically sending SIP option messages andmake the switch if the service provider fails to answer.Failover to secondary Internet service providerThe edge device should also be able to failover to a secondary Internet service providerif the primary goes down. It is important to be able to automatically switch back to theprimary once it is operational again. This will make it possible to have a cheaper backupInternet service provider.It should be noted that many service providers share the last mile, so there is really nopoint in having multiple service providers if they all use the same equipment.Ideally, the different connections should be divided (e.g. the primary Internet connectiondelivered with an optical fiber and the secondary as an xDSL line).Failover to secondary edge equipmentFinally, it should be possible to install the edge device in failover pair so it can failoverto a secondary unit if the primary experiences a hardware error. The type of failovercapability depends on the individual needs of the enterprise and can be divided intothree levels:1..Plain hardware failover where both registrations and on-going calls are lost.2..Failover with registrations maintained.3..Failover with both registrations and ongoing calls maintained.To make it meaningful to have a failover unit in a SIP trunking scenario, the unit should atleast have failover with registrations maintained, since with level one (above) it may taketime for the phones to realize that they need to re-register and thus it will take time tobecome operational again. It might be acceptable losing the ongoing call in the case ofa hardware failure, but it must be possible to redial again immediately when the failoverunit is activated.7 Implementing SIP trunking with ShoreTelShoreTel’s IP Telephony solutions provide robust and flexible deployment options for SIPtrunking. By partnering with Ingate Systems as a provider of Session Border Controllers(edge device), ShoreTel customers can count on highly secure SIP trunking, which isvalidated with a growing list of Internet Service Providers (ITSPs) for feature compatibilityand interoperability.ShoreTel and Ingate’s goal is to provide the customer peace of mind that the solution hasbeen tested and will be straightforward to deploy and maintain.
  • WORLD HEADQUARTERS 960 Stewart Drive, Sunnyvale, CA 94085 USA. shoretel.com and shoretelsky.com +1 (800) 425-9385 Toll Free +1 (408) 331-3300 Tel. +1 (408) 331-3333 Fax for ShoreTel +1 (646) 230-5000 Tel. +1 (646) 230-5001 Fax for ShoreTel Sky EMEA +800 408 33133 Freephone +44 (1628) 826300 Tel. ASIA PACIFIC +61 (0)2 9959 8000 Tel.Copyright © 2012 ShoreTel. All rights reserved. The ShoreTel logo and ShoreTel are registered trademarks of ShoreTel, Inc. in the United States and/or other countries. All other copyrights and trademarks hereinare the property of their respective owners. Specifications are subject to change without notice. Part #850-1354-01/09.12About ShoreTelShoreTel is a provider of business communication solutions whose brilliantly simple unifiedcommunications platforms, applications and mobile UC solutions promise a new rhythm ofworkforce engagement and collaboration. With costly complexity eliminated by design fromits award winning, all-in-one IP phone system, UC and contact center solution, and its industryleading hosted business phone system, workers enjoy a freedom and self-reliance that otherproviders can’t match. Users have full control to engage and collaborate, no matter the time,place or device, for the lowest cost and demand on IT resources in the industry. ShoreTel isheadquartered in Sunnyvale, California, and has regional offices and partners worldwide.For more information, visit shoretel.com or shoretelsky.com8 SummaryIn the end, it all boils down to this: is SIP trunking ready for prime time? The answeris yes. Indeed we can trust SIP trunking and its applications as long as we employ theright measures to secure media, ensure interoperability/future proof the network withstandards-based equipment, and are smart about the way SIP trunks are deployed.By including a SIP-capable edge device as part of the deployment, security, QoS andinteroperability issues can be reduced significantly. This translates into excellent voicequality, an easier deployment and seamless interoperability –a better overall experience.We see SIP trunks as paving the way to an all IP, all SIP world where businesses canwork without geographical constraints, employees can contribute equally regardless oflocation, and everyone is reachable anywhere and anytime as long as there is access toan Internet connection. This is the vision the IETF had when they first introduced the SIPprotocol, the idea of true global connectivity. SIP trunking extends the notion of seamlessconnectivity within a business to customers, remote employees, anyone working outsidethe corporate network. This is the next evolution of telecommunications – we lookforward to sharing it with you.ReferencesIETF SIP SpecificationsThere are numerous IETF RFCsand drafts that together definethe SIP standard. Rather thanlisting them all here we havechosen to point to an excellentsummary , provided by the IETF,called, “A Hitchhikers Guide toSIP.”This summary can be found here:http://tools.ietf.org/html/draft-ietf-sip-hitchhikers-guide-06The SIP Connect SpecSIPconnect relatedspecifications:sf-draft-twg-IP_PBX_SP_Interop-sibley-sipconnect