SIP Trunking in Dallas / Fort Worth TX for VoIP Systems and Internet
SIP Trunking & VoIP
What is SIP?
SIP stands for Session Initiation Protocol. It acts as a negotiator between multi-media protocols and initiates a session with the negotiated protocols between two endpoints. It can also implement new protocols or end a protocol during the session.
What is VoIP?
VoIP is simply using IP prtocols to send voice information bwtween two or more endpoints via an internal network or the internet.
Is a VoIP system expensive?
While there is an initial expense to purchase or lease new equipment, the monthly cost savings more than offsets the upgrade expenditure, and the contiued cost savings month to month compounds the advantages of a VoIP systems.
What is a PBX?
A PBX is a Private Branch Exchange. A PBX manages connection between extensions on a private network.
General Questions
Will I get the same features that I have with my traditional phone system?
Not only do you get the same basic features of your traditional phone system, you get many features you would normally have to pay for ( call waiting, call forwarding, hunt grouping, automatic attendant)
What is a DID?
A DID is a Direct Inward Dialing number assigned to a Communications Gateway and connected by Trunk to the PSTN ( Public Switched Telephone Network)
What is Digital Phone Service?
Digital Phone Service is Voice over Internet Protocol, but it has come to stand for new telephony technologies that use the internet to carry telephone signals. In the simplest terms it means making telephone calls over the internet.
How does digital voice quality sound?
In the early days of VoIP telephony, quality was a serious issue and early adopters needed high-speed connections and special software and hardware. Recent advances in available bandwidth and in IP telephony technology have meant that quality is only an issue for those in areas with limited bandwidth (internet connection speed). A recent study by Infotech indicates that quality and reliability of IP telephony solutions have met or exceeded expectations in 80% of cases.
Why is Digital Phone Important?
Digital Phone, Internet telephony and Pearl networks are all about bringing voice technologies like the telephone and data technologies like the internet together to provide greater benefits to users. Most advertising and promotion for Digital Phone service stresses cost savings, both for consumers and users of the services and for maintenance and operations for suppliers. But in fact, the real impact of Digital Phone service will come from the convergence of voice and data in the form of new features and capabilities. A simple example is the ability to automatically forward voicemail via email. Industry analysts predict an explosion in applications and capabilities around Digital Phone service over the next five years.
Switching Phone Services
What steps should an enterprise take to prepare for a Digital Voice Deployment?
Enterprises should be certain to complete several steps before beginning a Digital Voice deployment. These include developing a business case, specifying detailed requirements, testing the network, managing the vendor selection process, and launching a pilot project.
What constitutes a compelling Digital Voice business case?
One of the larger risks associated with Digital Voice is that it’s very much a bandwagon technology and everyone is jumping on. Every enterprise should make sure it has established a substantive business case for a Digital Voice migration. Cost savings are often important to the business justification (in terms of both toll bypass savings and management/infrastructure savings), but understanding required features and lifecycle requirements can offer a more compelling justification. For example, it may be critical that the company’s sales force have integrated messaging that allows them to check voicemail via email. Companies should also understand when they can end of life their existing telephony infrastructure, particularly their PBX.
What requirements should an enterprise specify in advance of a Digital Voice Deployment?
Companies should specify quite a few requirements that generally fall into three categories. First, traffic and service level related requirements are critical. These include metrics such as average and maximum number of simultaneous calls to run over network. Second, feature requirements that will drive usage by business units are essential to justifying the project at the executive level. For example, the customer service organization may require click to call capabilities because it reduces per call response times by 25 seconds. Third, specify the IT requirements necessary to support the aforementioned business objectives. This is critical to rationalizing procurement decisions that will be made later in the process.
Many buyers believe that Digital Voice is just another application running on the network. Not so. Digital Voice will tax and strain the network in ways that no enterprise can foresee. As a result, it is important to test the network in anticipation of a Digital Voice deployment. Numerous vendors now offer Digital Voice specific testing tools that can anticipate Digital Voice specific issues such as packet loss and latency, but IT departments must allocate time and brainpower to this effort. Understanding bandwidth, memory, and CPU requirements is critical to a successful deployment.
How should a company manage the vendor selection?
Adequate vendor selection depends on evaluating multiple vendors. It takes more time, but makes for a much better end result. Multiple vendor analysis can either be conducted through a formal RFP process or through less formal information collection techniques. Either way, score each vendor against a common set of criteria. From here, shortlist two to three vendors and begin negotiations. Key areas of negotiation include pricing, service levels, implementation schedules, and custom development.
Should enterprises engage in a Digital Voice pilot project?
Yes. Digital Voice deployments can and should be staged. Carving out a segment of users or specific office often allows IT groups to uncover pitfalls. Be sure to compare the pilot results against baseline results compiled from the legacy telephony solution. Examine network usage patterns and bottlenecks. Companies should also conduct user satisfaction to understand areas for improvement.
Network Monitoring & Testing
What do I need to do to make sure my Digital Voice implementation is a success?
Although a Digital Voice deployment can seem like a daunting task, achieving perfect voice quality essentially boils down to one thing: building a fault-free network with enough bandwidth to support your Digital Voice application. Testing and monitoring during the pre-deployment, implementation, and operation phases are the methods through which you verify that you have indeed created that network. If you have, and if you build in the ability to maintain that level no matter how your network changes, your Digital Voice deployment will be a success.
What do Testing and Monitoring mean?
Testing and monitoring of Digital Voice networks are two distinct but equally necessary components of a Digital Voice roll-out and of successful network management. A successful Digital Voice deployment depends significantly upon three factors: the capabilities of your existing network, the correct performance of your new system's functions, and the ability to receive real-time feedback regarding the status of calls on your network. These factors address the questions: Will it work? Does it work? and Is it continuing to work? Testing and monitoring solutions give you the tools to evaluate your strengths and liabilities in each of those areas.
What exactly is testing and how does it work?
Testing solutions usually test two things: the integrity of a network and the functionality of features. The solutions most often involve a hardware or software based signal generation tool paired with a diagnostic monitoring solution. The signal generation components manufacture call scenarios which are then evaluated by monitoring systems to determine how the system performs under conditions ranging from ordinary to extreme. This data is then compiled and, in a good system, highlights the areas of the network or system that are not functioning correctly or that need to be upgraded.
Digital Voice monitoring solutions provide feedback on the performance of the network with respect to actual calls. They can be either hardware or software based and often have the ability to report problems in real-time. Monitoring system feedback simplifies the process of troubleshooting the network and allows for greater network viability.
What is the difference between testing and monitoring?
Digital Voice testing products generate traffic in order to check specific Digital Voice network capabilities (max load functionality etc) at the time of testing. Because these products generate their own calls, they are indifferent to Digital Voice protocols. This makes them ideal for pre-deployment testing of the capabilities of the existing network. In addition to these capabilities, some vendors also verify specific Digital Voice features (call waiting etc) once the Digital Voice system has been installed.
Both Testing and Monitoring are necessary components to a successful Digital Voice deployment
Doesn't my service provider do this?
In most cases the answer is no, often service providers do not. According to a survey performed by Empirix, only 20% of service providers reported using Digital Voice application monitoring systems to ensure Digital Voice call quality.
Digital Voice analysis tools are often oriented toward particular Digital Voice protocol environments. As such, it is important to make sure that your testing solution is compatible with your particular protocol specifications.
Will testing products affect my system?
Traffic simulators embed hardware or software within the network and create simulated digital voice traffic across important paths. That traffic is then analyzed to provide feedback regarding the state of the network. Because traffic simulators travel on the network, they can be intrusive, especially when testing with multiple voice streams. This can potentially cause service issues in an active converged network.
Savings and Productivity
What are the primary savings benefits that my company can expect from Digital Voice?There are a number of areas where businesses can expect to save money by migrating to Digital Voice. These include start up savings, long distance savings, network infrastructure savings, conferencing savings, and productivity benefits or savings.
What types of capital savings can a company expect from Digital Voice?
When migrating to Digital Voice, companies can choose to deploy via a hosted solution provided by a service provider where the capital costs are minimal if any. Some service providers do charge for phones and small pieces of equipment such as wiring and adaptors. Pearl strives to make your return on investment (ROI) at the most three months.
How much should Digital Voice save on long distance?
Most of the savings associated with long distance is related to international calls. Where Digital Voice comes into play on LD is with regards to decreasing your inbound and outbound calling. Contact a Pearl rep to find out more. International calls remain to be very expensive and Digital Voice may save businesses as much as 40%.
Digital Voice delivers substantial savings on both audio and video conferencing. Audio-conference rates on traditional PSTN phone networks run 15 to 20 cents per minute on average. With Digital Voice, these rates may drop as low as 5 to 10 cents per minute. Video-conference rates run even higher with companies spending over $1 per minute in the legacy telephony world.
What type of network infrastructure savings can we expect?
Digital Voice runs on the IP network which currently supports data communications. As a result, there's no need to purchase or lease expensive PBX and dedicated networking equipment. Moreover; running Digital Voice on the IP network results in lower management and support costs. For example, adding new users or office locations to an IP network is generally a lower cost endeavor than adding them to a traditional PBX system. There are other smaller areas for savings benefits such as lower wiring costs and easier wireless rollouts that are the result of running voice and data on a single IP network.
When deploying a Digital Voice solution, the ownership and management of the data and voice networks usually converge into one group because a single network now supports voice and data communications. Legacy phone systems run on their own dedicated network with unique technologies, staffs, and processes required to run and support them. An additional area for productivity gains comes in the realm of move/add/change requests. Traditional PBX based move, add, or change requests can cost as much as $40 on average, but with Digital Voice, this average cost drops to below $10. Improved mobility for employees also pays savings dividends by allowing employees to log in to any phone and bring their number with them. This allows for more flexible work schedules that may improve worker productivity.
Digital Phone Service Level
What is Digital Voice Quality of Service (QoS)?Quality of Service or QoS is the quality of a call over a network. It also refers to the ability to prioritize certain types of traffic on an IP network. In the case of Digital Voice, this typically means prioritizing voice traffic at a higher level than other forms of traffic such as data so that voice traffic will not be delayed or dropped. Most QoS solutions focus on either resource reservation or resource prioritization.
What is Digital Voice latency?
Digital Voice latency causes delays in packet delivery. Physical distance, the number of router hops, encryption, and voice/data conversion all have an impact on latency. Users begin noticing latency as a service level issue when roundtrip latency is greater than 250 milliseconds (ms). The International Telecommunications Union recommends that latency never exceed 150 ms one way (from speaker to listener).
Digital Voice jitter occurs when voice packets are sent and received with timing variations. Jitter is effectively a variation of packet delay where delays actually impact the quality of the conversation. Think of jitter as variable delays in packet delivery. Participants will notice delays in the conversations impacted by jitter. As a result, many service providers now account for maximum jitter levels.
What is Digital Voice packet loss?
Digital Voice packet loss takes place when a large amount of traffic hits the network and causes it to drop packets. It usually manifests itself as dropped conversations or “tinny” sounds. Packet loss should never exceed 1% and most service providers guarantee service levels with .5% or less packet loss. Packet losses of 1% translates into one voice clip or skip every three minutes, while packet loss of .25% translates into one error every 53 minutes.
Latency and jitter really go hand in hand in most Digital Voice deployments. To effectively manage both, administrators should focus on reducing delay at the network endpoint and prioritizing traffic over the network. Optimizations of jitter buffering and packet size are good first steps to improving service quality at the endpoint. Endpoint delays can also be reduced by adhering to a standard packet size, using the G.711 codec, and avoiding asynchronous transcoding.
Prioritizing Digital Voice traffic over the network at Layers 2 and 3 also yields latency and jitter improvements. Policy based network management, bandwidth reservation, Type of Service, Class of Service, and Multi-Protocol Label Switching (MPLS) are all widely used techniques for prioritizing Digital Voice traffic at Layers 2 and 3.
Various solutions are available for Digital Voice packet loss. These include: packet loss concealment which hides the audible effects of lost packets; root cause analysis of individual switches and routers along the network path that may be the sole cause of loss; and, when packet loss manifests itself as jitter, monitoring for congestion.
IP-PBX FAQ
What is Digital Voice and how does it work?Digital Voice is VoIP which stands for “Voice over Internet Protocol.” It means voice communication over the Internet rather than over the traditional publicly switched telephone network (PSTN). In the same way that your computer turns your keyboard typing into an e-mail and transmits it via the Internet Protocol (IP) standard, it can also turn voice data into a form that is transmitted via IP and reassembled at the receiver’s computer or, increasingly, a specially equipped landline or mobile phone. By 2010, Gartner also predicts that IP-telephony products will represent 90 percent of new phone system sales. Traditional PBX systems are clearly in decline as customers opt for more open, feature rich, hybrid IP PBX solutions.
What is IP PBX?
An IP PBX takes the place of the PBX you may already have for your company’s PSTN calls. Like its PSTN cousin, an IP PBX (for private branch exchange) is an electronic switchboard that receives, routes, holds, forwards to voice mail, or otherwise manipulates calls that arrive over the Internet, rather than via the PSTN. It may be fully automatic or have a human receptionist who routes incoming calls from a main IP phone number to internal IP numbers or extensions. Where a PSTN PBX can connect many incoming and internal phone lines through a set of mechanical or electronic switches, an IP PBX will be mechanically simpler, typically either software that resides on a server or a small, independent server that connects with your existing data network.
What are the advantages of IP PBX?
An IP PBX provides more efficient switching and a more professional “look” than if everyone in a business has their own separate IP connection and account. It allows phone calls to be forwarded within the company and for internal voice-mail and conferencing capabilities that might otherwise have to be outsourced. An IP PBX is also much more flexible than a PSTN PBX, allowing an essentially infinite number of extensions and voice-mail boxes, plus desktop management via a Web browser rather than at a set of PSTN switches. They can also enable the recording of incoming and outgoing conversations (subject to legal considerations). IP PBXs, both as software and as physical devices, are relatively inexpensive and can be had for as little as $1,000.
Hosted IP PBX is a service provided off-site by a third-party company; with it, the IP PBX, associated equipment, and the responsibility for maintaining and upgrading the PBX lie with the hosted IP PBX provider.
How can our company start using an IP PBX?
Contact a Pearl representative that can assess your business’s needs and help you choose among the many companies that offer IP PBX products, including Alcatel, Avaya , Cisco , Dialexia, Digium, Fonality , Mitel, Nortel, Pingtel, Shoretel, Siemens and Talkswitch.
Why is the legacy PBX market in decline?
The emergence of open standards and protocols has created a new model in telephony where customers are increasingly in control. More recently, open source technology, such as Asterisk, has driven real technology innovation and parity between vendor and buyer when it comes to economic relationships. In the legacy market, PBX phone system vendors thrived in a world where they could sell proprietary systems, drive customers to single vendor environments, create customer lock-in via proprietary call control and endpoints, and force high margin phone and system deals on to customers.
What are the things that affect my Digital Voice Quality of Service (QoS)?
A variety of problems can affect Digital Voice Quality of Service, including dropped packets (too much data arrives at the receiving server too quickly), packet delay (data takes the long way around the Internet), jitter (different packets reach the receiver at different times) and related out-of-order delivery, and other mishandling of the data packets themselves. Each problem causes delays, which in time-sensitive settings like voice leads to lowered voice quality or even the dropping of whole calls.
Latency is the length of time it takes for your words to be received by a listener at the other end of a phone connection, typically expressed in milliseconds. According to a white paper from Brooktrout Technology, latency starts to affect phone conversations when it exceeds 150 milliseconds each way, and is unacceptable when it exceeds 450 milliseconds (nearly half a second). The company recommends engineering a Digital Voice system so that latency is always below 200 milliseconds and suggests some ways to accomplish this task here.
What are some ways to measure the performance of a Digital Voice network?
Latency is the total of a lot of small delays that occur during the coding, transmission and decoding of a voice conversation. The time each step takes can be measured, and then improved through better network equipment and better configuration of the network. Performance factors within a business’s control include the presence of absence of a firewall for Digital Voice traffic; delays in the digital signal processor (DSP) that codes and decodes voice traffic; how big a chunk of data the DSP takes on at once; how big a buffer you build in against delays elsewhere in the Digital Voice system; and how efficiently the decoded signal is routed within your data network to an IP-enabled phone or PC.
What is Asterisk?
In the same way that Linux is an open-source computer operating system, Asterisk is an open-source IP PBX. Like Linux, Asterisk has an enthusiastic developer community committed to improving it. It is available on a large number of platforms, including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. Asterisk is free, can interoperate between Digital Voice and traditional PSTN systems, and supports the protocols these two systems use, including H.323 (a PSTN protocol), Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).
What is SIP?
Session Initiation Protocol (SIP) is a set of standards that helps create, change and end IP sessions between one or more users, such as in the case of Digital Voice calls or audio conferences. It is notably lightweight and transport-independent, and was designed by the IP-community programmers rather than by the telecom industry.
Small-Medium Business
What are the benefits of Digital voice for business?Low cost is Digital Voice's primary benefit; if you have broadband Internet access, you already have the capacity to make digital voice calls from your computer without spending another dime, and you can eliminate most of your PSTN long-distance bill. Simplicity is a second benefit since voice and data can be handled by one set of network protocols and wiring. Purchasing a Digital Voice PBX system (including any dedicated software and hardware, and VoIP-enabled handsets) is competitive with updating a legacy PSTN system. In addition, using a computer to make Digital Voice calls means never being out of touch anywhere there is Internet access; employees can work from anywhere that has a fast, stable Internet connection.
Are there any disadvantages of Digital Voice for business?
Significant numbers of Digital Voice calls will tend to strain your internal data network and broadband Internet connection. Deploying Digital Voice on an industrial scale will likely require more Internet bandwidth, faster connections within your data network, and assigning different priorities to different types of Internet traffic (for instance, Digital Voice traffic should be given a higher priority than e-mail traffic in many cases, simply because voice is more time-sensitive. Also, by relying on Digital Voice and canceling commercial PSTN phone service, you are now at the mercy of your Internet service provider (ISP), and youthful ISPs are not as reliable as the PSTN is after its century of development.
What features does a small business need with Digital Voice?
Small businesses considering Digital Voice can get a surprising amount of features for very little cost. They will likely need the same features they currently have via PSTN: extension dialing, an auto attendant to answer the phone and route calls to Digital extensions, voice-mail boxes and audio conferencing. These seem to be available for most small-business Digital Voice packages and can even be handled by open-source IP PBXs such as Asterisk. VoIP-enabled mobile phones add cost, however.
What type of Internet connection does Digital Voice require?
While it’s possible to use a dial-up connection for Digital Voice calling, this connection speed—typically 56 kilobits per second (Kbps)—is too slow for commercial use. Business will want to have a broadband Internet connection that provides speeds of at least 256 Kbps (speeds of up to 30 megabits per second [Mbps] are available). Pearl Network can provide at least broadband speeds with the installation of a simple modem, but higher speeds may require dedicated equipment and always-on connections, which Pearl Network also provides.
What are all the pieces I need for Digital Voice (hardware, software, services, etc.)?
Businesses can implement Digital Voice between computers using nothing more than downloadable software like Pearl PC but for additional features (voicemail and audio conferences of more than 10 people, for example) you’ll want to add dedicated, for-pay business services such as Pearl Virtual PBX services or have Pearl provide an enterprise-grade IP PBX. Hardware requirements also vary but typically include headsets, IP-enabled phones, and a physical IP PBX box, if desired.
Can I use my company’s existing phone equipment and PBX with Digital Voice?
With additional special software and/or equipment, yes. For example, the Asterisk software-based IP PBX is designed to
work with PSTN and IP calls. Other third-party products like Digital Voice gateways can handle both SIP and H.323 (PSTN) phone traffic.
How should my business plan to migrate to Digital Voice?
Just as it would to a new PSTN system: Consider whether you need Digital Voice or whether your PSTN system is adequate to your business’s tasks (for now); the pace of change in both Digital Voice software and hardware; and the potential for changes in the regulatory environment, which could add taxes and thus costs to Digital Voice.
What is a Digital Voice gateway and do I need one?
A Digital Voice gateway is important for any reasonable-size Digital Voice deployment because it handles the task of translating and switching voice and fax traffic from a traditional PBX to IP form and back. They are available in either software or hardware form, with the advantage of stand-alone hardware being that, as a small computer and router in its own right, the gateway doesn’t rely on your data network’s processing power. Larger, enterprise-class gateways can handle this switching-and-translating task for up to thousands of voice channels.
Not on its own, since fax and voice use different transmission standards. However, there are a large number of vendors, such as FaxBack [link to vendor profile] and Intelliverse [link] whose software products bridge the fax/voice gap. Businesses can also maintain a dedicated PSTN fax line if they like, or use any of the many available efax software clients like J2 Messenger that send faxes as e-mail attachments.
Can Digital Voice handle a toll-free line?
Yes. Ask your service provider for details on how they support this.
Wireless Digital Voice
What is Wireless Digital Voice?
Wireless Digital Voice is Digital Voice running over a Wireless LAN (WLAN). These WLANs are typically compliant with the 802.11 standard. As long as callers are within range of a WLAN access point and using a Digital Voice enabled handset, they can make and receive calls over the wireless network. Wireless Digital Voice is gaining adoption in certain vertical industries, such as health care and retail, where worker mobility is critical to a productive workforce.
What are the shortcomings of Wireless Digital Voice today?
Traditionally, four factors have hindered the adoption of Wireless Digital Voice. First, the 802.11 standard poses scalability challenges for enterprise class Digital Voice deployments. Second, Quality of Service (QoS) has been lacking in wireless networks. The 802.11(e) version of the standard is specified to include QoS support. Third, implementing and maintaining a Wireless Digital Voice solution can be expensive and time consuming. Fourth, fast roaming is required for Wireless Digital Voice to operate in a seamless manner, something promised by the 802.11(r) standard.
Are there any security concerns with Wireless Digital Voice?
The exact same security issues that exist in the traditional Digital Voice world exist in the world of Wireless Digital Voice. These include risks such as eavesdropping and spam. There are additional security concerns that need to be addressed with Wireless Digital Voice primarily concerning the need to adequately secure the wireless portion of the network. If left unsecure, wireless communications can be intercepted by unauthorized third parties. These third parties can also gain access to the corporate network and may masquerade as valid access points, in the process duping users into entering the network via an invalid access point.
What is Unlicensed Mobile Access (UMA)?
The UMA specification was developed by wireless companies in an effort to retain control over cellular and Wi-Fi traffic. Unlicensed Mobile Access integrates Wi-Fi voice services with the mobile operators GSM networks. Subscribers with a Wi-Fi enabled cellular phone can switch from the network they use when out of the their home to Wi-Fi when they enter their home or the area covered by their Wi-Fi home network.
Digital Voice Security
What is Digital Voice eavesdropping?
Eavesdropping on Digital Voice calls takes place when unauthorized third parties monitor call signal packets. By eavesdropping, third parties can learn user names, passwords, and phone numbers thereby gaining control over calling plans, voicemail, call forwarding, and billing information. More importantly, third parties may also gain access to confidential business and personal information by eavesdropping on actual Digital Voice based conversations.
What is Digital Voice identity theft?
Digital Voice identity theft occurs when an unauthorized third party obtains a
user name and password and uses it to place calls at the expense of the legitimate account owner. In some cases, the third party may mislead other call participants into believing that the unauthorized third party is in fact the legitimate account owner.
What is a Digital Voice denial of service attack?
Digital Voice denial of service attacks (DoS attacks) overwhelms IP telephony devices with call requests and registrations. This flooding can create resource exhaustion, long term busy signals, and force disconnects of in session calls.
What is SPIT or Spam Over Internet Telephony?
Digital Voice spam is a relatively new threat with very few incidents reported thus far. Even so, every Digital Voice account has an IP address associated with it, allowing spammers to target thousands of IP addresses with messages. Most of these messages will end up clogging voicemail boxes, creating a need for more voice storage capacity and efficient voicemail management tools.
Phishing in the Digital Voice world is similar to Digital Voice SPIT except that the voice mail left for the account owner is purportedly from a trustworthy person or business and is designed to acquire sensitive information such as passwords or credit card numbers. Recipients may be provided a phone number or web address masquerading as a legitimate bank or online payment service.
What is Digital Voice service theft?
Similar in nature to PSTN based toll fraud, service theft in the Digital Voice world involves an unauthorized third party initiating outbound calls using a valid user’s name and password. Service fraud may also occur when a third party gains unauthorized physical access to a Digital Voice device. Typically, Digital Voice service theft is used for international calls where rates are higher.
What precautions should be taken against Digital Voice security threats?
First, ensure that all Digital Voice traffic is encrypted. There are multiple options here including VPNs and SRTP (Secure RTP), but make sure that the selected encryption method is efficient and fast. Otherwise, performance and throughput may be negatively impacted. Second, actively monitor for unauthorized or non-compliant technologies supporting the Digital Voice network. This includes identifying devices with non-standard configurations. Third, make Digital Voice servers physically secure by adopting technologies such as firewalls and intrusion detection. Be sure to use firewalls that can handle the unique attributes of Digital Voice traffic. Finally, require all users to login to access the Digital Voice network. A Digital Voice handset should be treated no differently than a user’s computer where network access is governed by login and password.