Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2

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1 Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and manage. While easy to use, the variety of choices available for connecting a Response Point phone system to a telephony service provider network can be slightly confusing. This white paper can help simplify the decision-making process by explaining the considerations involved in choosing a type of voice service and a service provider. Additionally, this paper discusses some common troubleshooting scenarios that will help you identify and resolve issues related to connectivity and voice service. Contents Voice Service Options... 2 Analog Service Considerations... 3 VoIP Service Considerations... 4 Troubleshooting Provider Issues... 5 Troubleshooting Analog Service Issues... 6 Troubleshooting VoIP Service Issues... 7 Access Link Issues... 7 Jitter... 7 Latency... 8 Send us feedback 1

2 Packet Loss... 8 Outages... 8 Glossary... 8 Voice Service Options Response Point supports three kinds of voice service analog, VoIP, and digital. Hardware devices called "gateways" allow each voice service to work with the Voice over Internet Protocol (VoIP) technology that underlies the phone system. VoIP uses various Internet protocols to carry voice traffic across your network. Electronic packets deliver these signals to phone-system components so that they can share information with one another. The communication that is relayed between software and hardware enables you to make calls, issue voice commands, save configurations, check voic , and so forth. Analog service is transmitted through a consumer-grade hard-wire connection that uses the landline, phone jacks, and cables associated with traditional phone service. Telephone companies (telcos) provide this voice service through the public-switched telephone network (PSTN). With analog service, you will need to configure at least one analog gateway, which might be a detached or built-in device, depending on the hardware manufacturer. When an external call comes into the phone system, the analog gateway converts incoming analog signals to digital ones, and then sends this data to your IP phones. When you place an outgoing call, the gateway reverses the process by converting digital signals back to analog ones, and then sends this data to the PSTN over the landline. VoIP service is transmitted through a broadband Internet connection via DSL or a cable modem, offered by VoIP service providers. After you configure service with a provider, you can send VoIP calls over the Internet connection. Some VoIP service providers require an Internet connection only, with no additional hardware. Others may require a VoIP gateway device to help improve service quality, and to allow customer support specialists to remotely edit your VoIP account and refresh configuration data. Note A converter (sometimes called an "integrated access device") might have come with the VoIP service you purchased. The IAD translates between digital signals and analog signals. Although your phone company or other service provider recognizes this as "VoIP service," it appears to Response Point as "analog service" because the interface between Response Point and the IAD is an analog one. Digital service is transmitted through an enterprise-grade hard-wire connection. "T1" (in the U.S.), "E1" (in Europe),"PRI," or "ISDN" service providers will run a special cable into your office building, similar to the cabling installed by a telco. Digital service lines normally have up to 24 rotating channels that can deliver voice and data packets to phone-system components. You will need to configure a digital Send us feedback 2

3 gateway with this voice service. Some digital service providers may offer fractional digital service, which means you purchase only a portion of the circuit that may have only 8 available channels instead of 24. Analog Service Considerations Response Point is designed to use analog connections as a method of connecting to external telephony services. This type of installation will be most common when deploying Response Point solutions in small-business environments. When selecting an analog service provider, there are several things that you need to do to ensure that the Response Point system functions correctly with the analog service provider network. The following tasks should be completed prior to connecting a Response Point system to a PSTN network: Stick with well-known providers that use Bell-standard line configurations. Some smaller PSTN providers use incompatible standards that may prevent Response Point from functioning properly. Calculate the number of phone lines needed to support the business by counting the number of users and determining how many peak calls are normally expected. If the number of peak calls means that X connections will occur simultaneously, you must have X lines to ensure uninterrupted service. Calculate the number of dedicated lines needed to support analog devices such as fax machines and security systems that depend on analog phone service to function correctly. While some devices might be capable of using an ATA device to connect to a VoIP network, you will need to check with the manufacturer to make certain that ATA connections are supported. When planning a deployment date, check with the analog service provider to determine how long it takes to provision and activate the necessary lines. You will need to cushion the deployment date in order to have time to test that the additional lines will be fully functional when deployment occurs. In addition to checking lines for activation, also ensure that all requested features have been enabled on all lines. Some Response Point services depend on caller ID, so it is important to make sure that all lines have caller ID enabled. Response Point s Automated Receptionist feature reliably supports only 8 simultaneous speechrecognition connections. If the business expects more than 8 simultaneous inbound calls, it may Send us feedback 3

4 want to designate an employee to answer calls rather than rely on the Automated Receptionist feature. VoIP Service Considerations It is possible to use Response Point in conjunction with an IP telephony provider as part of a comprehensive, end-to-end VoIP solution. There are some obvious advantages to this approach, especially when connecting multiple sites to each other when a high-bandwidth WAN connection is in place, or when long-distance utilization is high. We recommend that you use a preferred VoIP service provider that has been verified to reliably support Response Point. However, Response Point uses standard protocols and may work with other service providers. At a minimum, your service provider must support: SIP signaling g.711 audio encoding Session border controller or application level gateway for NAT traversal Registration using the SIP REGISTER method Response Point will require the following information from the VoIP service provider: Service provider name SIP proxy or registration server address Registration interval User Address of Record (AOR) or Uniform Resource Identifier (URI) Default domain Authentication ID or SIP user ID Password Caller ID display name For best results, use a DSL or faster connection. More VoIP service considerations: Send us feedback 4

5 It is a common misconception that VoIP long-distance services are free or can be connected to any location. Ensure that your customers carefully review VoIP provider information to determine long-distance costs and capabilities. Before choosing a provider, consider the customer s typical outbound long-distance usage patterns and the service levels the customer will require. Ask the provider how many simultaneous calls its service can support; some providers can support only a limited number of simultaneous calls. Confirm that the provider can deliver caller ID services. Some Response Point features depend on caller ID to function correctly. Confirm that the VoIP service provider uses the SIPconnect interface specification. You will also need to determine whether the customer s firewall and router can allow and prioritize the service provider s traffic. Confirm that the current router supports QoS for voice traffic. If the router that will be used for VoIP traffic does not support QoS, the customer may need to purchase additional network equipment. Determine whether the current firewall can allow inbound and outbound VoIP traffic to pass through. Some less expensive or bundled firewalls may not allow you to specify the types of packets that may be allowed through in both directions. If the firewall does not support the provider s VoIP standard, the customer may need to purchase a new firewall. Ask the service provider to specify the firewall settings it will require for its service. Troubleshooting Provider Issues The first step in resolving problems in a Response Point environment is to determine the source of the problem. Typically, there are a few possible sources of problematic behavior in any VoIP environment, including Response Point environments. Generally, these problems can be broken into the following categories: Line quality Phone units Internal network cabling and patch cables Local area network (LAN) traffic congestion Send us feedback 5

6 PBX units Switch settings or hardware Router settings or hardware Firewall settings or hardware Link access problems External network traffic congestion VoIP call establishment delays Provider-based connectivity or setting problems While it is possible for you to directly isolate, troubleshoot, and resolve most of these problems, some of them are not due to anything that occurs on the Response Point system itself. This section will give you information that can be used to determine when problems are not a part of the Response Point system and may require the assistance of a service provider to resolve. Troubleshooting Analog Service Issues Generally speaking, the Response Point network effectively ends at the connection between the VoIP PBX and the PSTN analog lines. At this point the VoIP traffic has been converted to an analog signal and is carried by the analog service provider to its destination. Any issues that originate outside the base unit will likely require the assistance of the analog service provider to resolve. Some of the more common issues that originate with analog service providers include: Dead lines - Symptoms can include no dial tone, customers reporting frequent busy signals, nobody answering when the business places outgoing calls, and the inability to make as many simultaneous calls as the user should be able to make based on the unit s configuration. The Response Point administrative interface can be used in some cases to help spot problems caused by dead lines. However, these issues usually require the intervention of a service provider. You can reduce the risk of dead lines by testing all lines prior to Response Point deployment to ensure that they have been activated. Insufficient line-to-peak-demand ratio - Symptoms can be similar to dead line issues because calls will not go through when all lines are in use. When the number of simultaneous calls is greater than the Send us feedback 6

7 number of available lines, callers will receive a busy signal and users will be unable to place calls. When these problems start to occur, it may be time to purchase additional lines from the service provider. Features not working - Feature functionality issues are usually due to the provider failing to enable caller ID or rollover features on all lines connected to the Response Point base unit. Even seemingly unrelated features, including Response Point s direct-dial features, can be dependent on carrier services like caller ID. To prevent these problems, test the features available on all lines prior to deploying Response Point and whenever a new line is added to the Response Point system. Troubleshooting VoIP Service Issues This section provides some general guidelines for identifying VoIP service issues that may reside outside of the Response Point system. Access Link Issues Access link issues are problems that occur between local area networks and lower-bandwidth networks like Internet connections and WAN links. Access link congestion issues can cause a number of symptoms, including: Pauses or delays in conversations, or gaps in speech. When the router s buffer is full and it cannot send packets correctly, it can cause excessive pauses or delays in voice traffic that can, in turn, cause conversational difficulties. Random popping sounds or garbled speech. This can be caused by access link congestion as well, since routers will invoke random early detection when their buffers fill, dropping packets to prompt the sender to resend packets. When this occurs, voice packets can be lost or sent out of order, causing voice quality issues or popping sounds. Occasional ticking sounds, sometimes at regular intervals. This is usually caused by routers switching routes. This causes timing issues that result in a ticking-like background sound. Excessive echo or tunnel voice quality. While this problem can also occur because of poor analog line quality, an excessive echo during conversations or a hollow sounding voice can be caused by improper phone volume settings. Jitter Jitter describes problems associated with packet timing issues. Users will notice garbled voices or dropped portions of words. This is usually caused by problems along the network path, either due to congestion, route changes, or similar issues. These occur most frequently outside of the local area network, so troubleshooting should begin at the router and may involve the service provider s network. Send us feedback 7

8 Latency Latency refers to delays caused by the distance packets must travel between the router and the service provider s network, or when buffers are used to compensate for excessive jitter. Most major service providers have multiple network access points that are positioned to reduce the effect of latency. When the ping time from the local network to the service provider exceed 250ms, users may notice a significant delay between the time they speak and the time the person on the other end hears what was said. This can also result in excessive echoing or even tunnel voice quality. Packet Loss While the effect of lost packets or delayed packets on data traffic is usually unnoticed, it can cause major quality issues over voice networks. When packet loss is minor, users will notice an increase in echo or robot-like voice quality. High levels of packet loss or significant delays can cause speech to break up or become drastically distorted. Packet loss or delayed delivery is generally caused by network saturation anywhere along the network path, but generally only occurs on the local network when hubs are used or when computers are attached to the network through the phone s built-in switch. Outages Outages can be caused by hardware failures along the network path or by failed links along the network path. To isolate the source of network failures, perform pings or trace routes to locate the network failure. If the failure occurs outside of the local area network, you will need to contact the service provider to resolve the issue. Glossary ATA - An ATA (analog telephony adapter), also called an analog gateway device, converts standard analog RJ-11 connections to Ethernet RJ-45 connections so that analog devices can be used on VoIP networks. Automated Receptionist - The Automated Receptionist (or auto attendant) is a Response Point voicerecognition technology that can prompt callers for information and respond to their requests. DID - Direct Inward Dialing (DID) is a virtual phone number that can be dialed directly from outside of the business and will be forwarded to a specific VoIP phone. FXO - A Foreign Exchange Office (FXO) is the customer side of the connection between an external phone service provider (FXS or Foreign Exchange Station) that generates a ring signal and the customer (FXO) that receives a ring signal. Send us feedback 8

9 IAD - An Integrated Access Device (IAD) enables the conversion of analog and digital signals for convergence of network services. IP Telephony - This is the general term used to describe VoIP. ITSP - An Internet Telephony Service Provider (ITSP) is a company that offers Internet-based data service for VoIP telephony. PBX - A PBX (Private Branch Exchange) is a private telephone switch that provides full switching features for an office or campus. POTS - Plain Old Telephone Service (POTS) means simple analog telephone service. POTS is sometimes used interchangeably with PSTN. PSTN - Public Switched Telephone Network (PSTN) is the switched analog voice network that most people use for telephony today. QoS - Quality of Service (QoS) is a method used to mark and prioritize different types of network traffic passing through a gateway or router to enable time-sensitive traffic, such as voice conversations, to have transmission priority over data packets. RTP - Real-time Transport Protocol (RTP) is a standardized packet format for streaming multimedia content and VoIP communications. RTP utilizes Session Initiation Protocol (SIP) to initiate and terminate sessions. SIP - Session Initiation Protocol (SIP) is a transport-independent application layer control protocol often used by multimedia or VoIP applications to initiate, maintain, and terminate connection sessions with one or more participants. VoIP - Voice over Internet Protocol (VoIP) is the practice of transmitting voice communications over data networks by using Internet protocols. Send us feedback 9

10 2009 Microsoft Corporation. All rights reserved. Microsoft, Response Point, and Windows are trademarks of the Microsoft group of companies. The example companies, organizations, products, domain names, addresses, logos, people, places, and events depicted herein are fictitious. No association with any real company, organization, product, domain name, address, logo, person, places, or events is intended or should be inferred. Send us feedback 10

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