How to configure Voice gateway in Cisco

how to configure voice gateway in call manager and how to add voice gateway to callmanager how does voice gateway work configure linksys voice gateway with router
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Published Date:25-10-2017
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Introducing Voice Gateways After reading this chapter, you should be able to perform the following tasks: ■ Describe the characteristics and historical evolution of unified communications net- works, the three operational modes of gateways, their functions, and the related call leg types. ■ Explain how gateways route calls and which configuration elements relate to incom- ing and outgoing call legs. ■ Describe how to connect a gateway to traditional voice circuits using analog and digital interfaces. ■ Define DSPs and codecs, and explain different codec complexities and their usage. Cisco Unified Communications gateways play an important role in the Cisco Unified Communications environment. Their primary function is to convert voice formats, signals, and transmission methods as voice information travels over various network types. This chapter describes the various types of voice gateways and how to deploy them in different Cisco Unified Communications environments. Furthermore, it explains the call-routing process, the direct inward dialing (DID) feature, the various types of voice ports and their characteristics, coder-decoders (codecs), digital signal processors (DSP), and their implementation. The Role of Gateways This section describes the operational modes of a voice gateway and how the gateway fits in the Cisco Unified Communications architecture. It explains the voice gateway functions in each Cisco Unified Communications deployment model and the call legs that are associated with each operational mode.2 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide Traditional Telephony Networks The following components are common elements in such a telephony network, as shown in Figure 1-1. Edge Devices Interoffice Trunks San Jose CO CO Boston Tie Tie Trunks Trunks Switch Switch PBX PBX CO CO Trunks Trunks Local Local Loops Loops PSTN Figure 1-1 Traditional Telephony Network ■ Telephones: Analog telephones are the most common type of phone in a traditional telephony network. Analog phones directly connect to the public switched telephone network (PSTN). ■ Central office (CO) switch: These switches terminate the local loop and manage sig- naling, digit collection, call routing, call setup, and call teardown. ■ Private branch exchange (PBX): A PBX is a privately owned switch that is located on the customer premises. A PBX is a smaller, privately owned version of the CO switches that telephone companies (telcos) use. Many businesses still have a PBX telephone system. Large offices with more than 50 telephones or handsets still use a PBX to connect users, both in-house and to the PSTN. ■ Trunk: Trunks provide the path between two switches and can be of different types: ■ CO trunk: A CO trunk is a direct connection between a local CO and a PBX, which can be analog or digital. ■ Tie trunk: A tie trunk is a dedicated circuit that connects PBXs to each other.Chapter 1: Introducing Voice Gateways 3 ■ Interoffice trunk: An interoffice trunk is typically a digital circuit that connects the COs of two local telcos. Traditional telephony differs in many aspects from modern unified communications. One important difference is the closed nature of traditional telephony. Integration with mod- ern software applications, databases, and a rapidly evolving computing environment is difficult. Traditional telephony uses circuit-switching technology to establish a voice channel end to end. This approach does not allow sharing of the network infrastructure for emerging applications and services. A traditional telephony environment addresses these areas: ■ Signaling: Signaling is the ability to generate and exchange the control information that will be used to establish, monitor, and release connections between two end- points. Voice signaling requires the ability to provide supervisory, address, and alert- ing functionality between nodes. The PSTN network uses Signaling System 7 (SS7) to transport control messages. SS7 uses out-of-band signaling, which, in this case, is the exchange of call control information in a separate dedicated channel. ■ Database services: Database services include access to billing information, caller name (CNAM) delivery, toll-free database services, and calling-card services. An example is providing a call notification service that places outbound calls with prere- corded messages at specific times to notify users of such events as school closures, wakeup calls, or appointments. ■ Bearer control: Bearer control defines the bearer channels that carry voice calls. Proper supervision of these channels requires that the appropriate call connect and call disconnect signaling is passed between end devices. Correct signaling ensures that the channel is allocated to the current voice call and that the channel is properly deallocated when either side terminates the call. Connect and disconnect messages are carried by SS7 in the PSTN network. As you will learn in your continued unified communications studies, unified communica- tions solutions exist for signaling, database services, and bearer control. Cisco Unified Communications Overview The Cisco Unified Communications system fully integrates communications by enabling data, voice, and video to be transmitted over a single network infrastructure using stan- dards-based IP. The Cisco Unified Communications system incorporates and integrates the following communications technologies: ■ IP communications is the technology that transmits voice and video communications over a network using IP standards. Cisco Unified Communications includes hardware and software products, such as call-processing agents, IP phones (both wired and wireless), voice-messaging systems, video devices, and many special applications. ■ Mobile applications enhance access to enterprise resources, increase productivity, and increase the satisfaction of mobile users.4 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide ■ Customer care enables efficient and effective customer communications across a global network. This strategy allows organizations to draw from a broader range of resources to service customers. They include access to a large pool of agents and multiple channels of communication, as well as customer self-help tools. ■ Telepresence and conferencing enhance the virtual meeting environment with an inte- grated set of IP-based tools for voice, video, and web conferencing. ■ Messaging provides the functionality for sending and managing of voice and video messages for users. ■ Enterprise social software includes applications that enable communications with the enterprise that are not strictly limited to business-oriented activities. Cisco Unified Communications Architecture Leveraging the framework provided by Cisco IP hardware and software products, the Cisco Unified Communications system has the capability to address current and emerg- ing communications needs in the enterprise environment. The Cisco Unified Communications family of products is designed to optimize feature functionality, reduce configuration and maintenance requirements, and provide interoperability with a wide variety of other applications. The Cisco Unified Communications architecture, as illustrated in Figure 1-2, consists of these logical layers: Cisco Unified Wireless Unified Unified Personal IP Mobile Endpoints IP Phones IP Phones IP Phone 7985 Communicator Communicator Phones Unified Unified Unified Cisco Unity Unified Personal IP Mobile MeetingPlace Customer Video Applications Messaging Communicator Communicator Communicator Conferencing Contact Advantage Services Smart Business Unified CM Communications Cisco Unified Unified CM Unified Express Presence System Business Edition CM/SME/IME Infrastructure Routing Management QoS Security Switching Availability Administration Figure 1-2 Cisco Unified Communications ArchitectureChapter 1: Introducing Voice Gateways 5 ■ Infrastructure: Infrastructure consists of Cisco network components. It provides and maintains a high level of availability, quality of service (QoS), and security for the network. ■ Services: Services are responsible for providing the core functionality of Cisco Unified Communications, such as signaling and call routing. ■ Applications: Applications include a wide array of software that offers a collection of features to the users. ■ Endpoints: Endpoints include end-user hardware and software products that consti- tute attachment points to the Cisco Unified Communications system. Cisco Unified Communications Business Benefits The business advantages that influence the implementation of Cisco Unified Communications have changed over time. Starting with simple media convergence, these advantages have evolved to include call-switching intelligence and the total user experi- ence. Consider the following business drivers for a unified communications solution: ■ Cost savings: Traditional time-division multiplexing (TDM), which is used in the PSTN environment, dedicates 64 kbps of bandwidth per voice channel. This approach results in unused bandwidth when there is no voice traffic. VoIP shares bandwidth across multiple logical connections, which makes more efficient use of the band- width and therefore reduces bandwidth requirements. ■ Flexibility: The sophisticated functionality of IP networks allows organizations to be flexible in the types of applications and services that they provide to their cus- tomers and users. Service providers can easily segment customers. This segmentation helps them to provide different applications, custom services, and rates, depending on the traffic volume needs and other customer-specific factors. ■ Advanced features: Here are some examples of the advanced features provided by Cisco Unified Communications: ■ Advanced call routing: When multiple paths exist to connect a call to its destina- tion, some of these paths might be preferred over others based on cost, distance, quality, partner handoffs, traffic load, or various other considerations. Least-cost routing and time-of-day routing are two examples of advanced call routing that can be implemented to determine the best possible route for each call. ■ Unified messaging: Unified messaging improves communications and produc- tivity. It provides a single user interface for messages that have been delivered over various media. For example, users can read their email, hear their voice mail, and view fax messages by accessing a single inbox. ■ Integrated information systems: Organizations use Cisco Unified Communications to affect business process transformation. These processes include centralized call control, geographically dispersed virtual contact cen- ters, and access to resources and self-help tools.6 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide ■ Long-distance toll bypass: Long-distance toll bypass is an attractive solution for organizations that place a significant number of calls between sites that are charged traditional long-distance fees. In this case, it might be more cost effec- tive to use VoIP to place those calls across the IP network. If the IP WAN becomes congested, calls can overflow into the PSTN, ensuring that there is no degradation in voice quality. ■ Voice and video security: There are mechanisms in the IP network that ensure secure IP conversations. Encryption of sensitive signaling header fields and mes- sage bodies protects the packets in case of unauthorized packet interception. ■ Customer care: The ability to provide customer support through multiple media, such as telephone, chat, and email, builds solid customer satisfaction and loyalty. A pervasive IP network allows organizations to provide contact center agents with consolidated and up-to-date customer records along with the related customer communication. Access to this information allows quick problem solv- ing, which, in turn, builds strong customer relationships. ■ Telepresence and conferencing services: These services save time and resources by providing a media-rich communications platform for users in a distributed enterprise environment. Originally, return on investment (ROI) calculations centered on toll-bypass and converged network savings. Although these savings are still relevant today, advances in voice tech- nologies allow organizations and service providers to differentiate their product offerings by providing advanced features such as those in the preceding list. Cisco Unified Communications Gateways Unified communications gateways are connection points between different communica- tions networks. Depending on the deployment type, a gateway can perform one or sever- al of these functions: ■ Act as a voice switch that interconnects multiple traditional telephony circuits. The circuits can be analog or digital. The gateway participates in signaling and might have to convert the media channels. Gateways provide physical access for local analog and digital voice devices such as telephones, fax machines, key sets, and PBXs. ■ Act as a PSTN-to-VoIP gateway that provides translation between VoIP and non- VoIP networks, such as the PSTN. In addition to the functionality of traditional voice switches, the PSTN-to-IP gateways enable voice and video communications between traditional PSTN infrastructure and converged IP networks. ■ Act as a Cisco Unified Border Element (often written as Cisco UBE or CUBE) that in- terconnects two IP networks and allows communications between endpoints distrib- uted among them. The Cisco UBEs might implement filtering, address translation, and security-related functions.Chapter 1: Introducing Voice Gateways 7 Gateway Operation Cisco Unified Communications gateways use several control and call-signaling protocols. Among these protocols are ■ H.323: H.323 is a standard that specifies the components, protocols, and procedures that provide multimedia communication services and real-time audio, video, and data communications over packet networks, including IP networks. H.323 is part of a fam- ily of International Telecommunication Union Telecommunication Standardization sector (ITU-T) recommendations called H.32x that provides multimedia communica- tion services over a variety of networks. H.32x is an umbrella of standards that de- fines all aspects of synchronized voice, video, and data transmission. It also defines end-to-end call signaling. ■ Media Gateway Control Protocol (MGCP): MGCP is a method for PSTN gateway control or thin device control. Specified in RFC 2705, MGCP defines a protocol that controls VoIP gateways that are connected to external call control devices, referred to as call agents. MGCP provides the signaling capability for edge devices, such as gateways, that might not have implemented a full voice-signaling protocol such as H.323. For example, anytime an event, such as off-hook, occurs on a voice port of a gateway, the voice port reports that event to the call agent. The call agent then sig- nals the voice port to provide a service, such as dial-tone signaling. ■ Session Initiation Protocol (SIP): SIP is a detailed protocol that specifies the com- mands and responses to set up and tear down calls. SIP also details features such as security, proxy, and Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) services. SIP and its partner protocols, Session Announcement Protocol (SAP) and Session Description Protocol (SDP), provide announcements and information about multicast sessions to users on a network. SIP defines end-to-end call signaling between devices. SIP is a text-based protocol that borrows many elements of HTTP, using the same transaction request and response model and similar header and response codes. It also adopts a modified form of the URL addressing scheme used within email that is based on Simple Mail Transfer Protocol (SMTP). ■ Skinny Client Control Protocol (SCCP): SCCP is a Cisco proprietary protocol used between Cisco Unified Communications Manager and Cisco IP Phones. The end sta- tions (IP phones) that use SCCP are called Skinny clients, which consume less pro- cessing overhead. The client communicates with the Cisco Unified Communications Manager (often referred to as Call Manager, and abbreviated UCM) using connection- oriented (TCP-based) communication, which is sometimes used to establish a call with another H.323-compliant end station. The following sections describe each of these protocols in greater detail. www.allitebooks.com8 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide The H.323 Protocol Suite H.323 is a suite of protocols defined by the ITU for multimedia conferences over LANs. The H.323 protocol was designed by the ITU-T and was initially approved in February 1996. It was developed as a protocol that provides IP networks with traditional telephony functionality. Today, H.323 is the most widely deployed standards-based voice and video- conferencing standard for packet-switched networks. The protocols specified by H.323 include the following: ■ H.225 Call Signaling: H.225 call signaling is used to establish a connection between two H.323 endpoints. This is achieved by exchanging H.225 protocol messages on the call-signaling channel. The call-signaling channel is opened between two H.323 endpoints or between an endpoint and an H.323 gatekeeper. ■ H.225 Registration, Admission, and Status: Registration, admission, and status (RAS) is the protocol between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes, status, and disengage procedures between endpoints and gatekeepers. A RAS channel is used to exchange RAS messages. This signaling channel is opened between an end- point and a gatekeeper prior to the establishment of any other channels. ■ H.245 Control Signaling: H.245 control signaling is used to exchange end-to-end control messages governing the operation of an H.323 endpoint. These control mes- sages carry information related to the following: ■ Capabilities exchange ■ Opening and closing of logical channels used to carry media streams ■ Flow-control messages ■ General commands and indications ■ Audio codecs: An audio codec encodes the audio signal from a microphone for transmission by the transmitting H.323 terminal and decodes the received audio code that is sent to the speaker on the receiving H.323 terminal. Because audio is the minimum service provided by the H.323 standard, all H.323 terminals must have at least one audio codec supported, as specified in the ITU-T G.711 recommendation (coding audio at 64 kbps). Additional audio codec recommendations, such as G.722 (64, 56, and 48 kbps), G.723.1 (5.3 and 6.3 kbps), G.728 (16 kbps), and G.729 (8 kbps), might also be supported. ■ Video codecs: A video codec encodes video from a camera for transmission by the transmitting H.323 terminal and decodes the received video code on a video display of the receiving H.323 terminal. Because H.323 specifies support of video as op- tional, the support of video codecs is optional as well. However, any H.323 terminal providing video communications must support video encoding and decoding as spec- ified in the ITU-T H.261 recommendation.Chapter 1: Introducing Voice Gateways 9 In Cisco IP Communications environments, H.323 is widely used with gateways, gate- keepers, and third-party H.323 clients, such as video terminals. Connections can be con- figured between devices using static destination IP addresses. Note Because H.323 is a peer-to-peer protocol, H.323 gateways are not registered with Cisco Unified Communications Manager as an endpoint is. An IP address is configured in the Cisco UCM to direct calls to the H.323 device. MGCP MGCP is a client/server call control protocol built on a centralized control architecture. MGCP offers the advantage of centralized gateway administration and provides for large- ly scalable IP telephony solutions. All dial plan information resides on a separate call agent. The call agent, which controls the ports on the gateway, performs call control. An MGCP gateway does media translation between the PSTN and VoIP networks for exter- nal calls. In a Cisco-based network, Cisco Unified Communications Managers function as call agents. MGCP is a plain-text protocol used by call control devices to manage IP telephony gate- ways. MGCP was defined under RFC 2705, which was updated by RFC 3660, and super- seded by RFC 3435, which was updated by RFC 3661. With MGCP, Cisco UCM knows of and controls individual voice ports on an MGCP gateway. This approach allows complete control of a dial plan from Cisco UCM and gives Communications Manager per-port control of connections to the PSTN, legacy PBX, voice-mail systems, and plain old telephone service (POTS) phones. MGCP is implement- ed with use of a series of plain-text commands sent via User Datagram Protocol (UDP) port 2427 between the Cisco UCM and a gateway. Note that for an MGCP interaction to take place with Cisco UCM, an MGCP gateway must have Cisco UCM support. If you are a registered customer of the Software Advisor, you can use this tool to make sure your platform and your Cisco IOS software or Cisco Catalyst operating system version are compatible with Cisco UCM for MGCP. Also, make sure your version of Cisco UCM supports the gateway. A Primary Rate Interface (PRI) and Basic Rate Interface (BRI) backhaul is an internal interface between the call agent (such as Cisco UCM) and Cisco gateways. It is a separate channel for backhauling signaling information. A backhaul forwards PRI Layer 3 (Q.931) signaling information via a TCP connection. An MGCP gateway is relatively easy to configure. Because the call agent has all the call- routing intelligence, you do not need to configure the gateway with all the dial peers it would otherwise need. A downside is that a call agent must always be available. Cisco MGCP gateways can use Survivable Remote Site Telephony (SRST) and MGCP fallback to allow the H.323 protocol to take over and provide local call routing in the absence of a Communications Manager (for example, during a WAN outage). In that case, you must configure dial peers on the gateway for use by H.323.10 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide Session Initiation Protocol SIP is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999; RFC 3261, pub- lished in June 2002; and RFC 3665, published in December 2003. Because SIP is a com- mon standard based on the logic of the World Wide Web and is very simple to imple- ment, it is widely used with gateways and proxy servers within service provider networks for internal and end-customer signaling. SIP is a peer-to-peer protocol where user agents (UA) initiate sessions, similar to H.323. However, unlike H.323, SIP uses ASCII-text-based messages to communicate. Therefore, you can implement and troubleshoot SIP very easily. Because SIP is a peer-to-peer protocol, the Cisco UCM does not control SIP devices, and SIP gateways do not register with Cisco UCM. As with H.323 gateways, only the IP address is available on Cisco UCM to make communication between a Cisco UCM and a SIP voice gateway possible. Skinny Client Control Protocol SCCP is a Cisco proprietary protocol that is used for the communication between Cisco UCM and terminal endpoints. SCCP is a client/server protocol, meaning any event (such as on-hook, off-hook, or buttons pressed) causes a message to be sent to a Cisco UCM. Cisco UCM then sends specific instructions back to the device to tell it what to do about the event. Therefore, each press on a phone button causes data traffic between Cisco UCM and the terminal endpoint. SCCP is widely used with Cisco IP Phones. The major advantage of SCCP within Cisco UCM networks is its proprietary nature, which allows you to make quick changes to the protocol and add features and functionality. SCCP is a simplified protocol used in VoIP networks. Cisco IP Phones that use SCCP can coexist in an H.323 environment. When used with Cisco Communications Manager, an SCCP client can interoperate with H.323-compliant terminals. Comparing VoIP Signaling Protocols The primary goal for all four of the previously mentioned VoIP signaling protocols is the same—to create a bidirectional Real-time Transport Protocol (RTP) stream between VoIP endpoints involved in a conversation. However, VoIP signaling protocols use different architectures and procedures to achieve this goal. H.323 H.323 is considered a peer-to-peer protocol, although H.323 is not a single protocol. Rather, it is a suite of protocols. The necessary gateway configuration is relatively com- plex, because you need to define the dial plan and route patterns directly on the gateway. Examples of H.323-capable devices are the Cisco VG224 Analog Phone Gateway and the Cisco 2600XM Series, Cisco 2800 Series, 2900 Series, and 3900 Series routers.Chapter 1: Introducing Voice Gateways 11 The H.323 protocol is responsible for all the signaling between a Cisco UCM cluster and an H.323 gateway. The ISDN protocols, Q.921 and Q.931, are used only on the Integrated Services Digital Network (ISDN) link to the PSTN, as illustrated in Figure 1-3. PSTN V H.323 Q.921 Q.931 Figure 1-3 H.323 Signaling MGCP The MGCP protocol is based on a client/server architecture. That simplifies the configu- ration because the dial plan and route patterns are defined directly on a Cisco UCM serv- er within a cluster. Examples of MGCP-capable devices are the Cisco VG224 Analog Phone Gateway and the Cisco 2600XM Series, 2800 Series, 2900 Series, and 3900 Series routers. Non-IOS MGCP gateways include the Cisco Catalyst 6608-E1 and Catalyst 6608-T1 module. MGCP is used to manage a gateway. All ISDN Layer 3 information is backhauled to a Cisco UCM server. Only the ISDN Layer 2 information (Q.921) is terminated on the gate- way, as depicted in Figure 1-4. PSTN V MGCP Q.921 Q.931 Figure 1-4 MGCP Signaling SIP Like the H.323 protocol, SIP is a peer-to-peer protocol. The configuration necessary for the gateway is relatively complex because the dial plan and route patterns need to be defined directly on the gateway. Examples of SIP-capable devices are the Cisco 2800 Series, 2900 Series, and 3900 Series routers. The SIP protocol is responsible for all the signaling between a Cisco UCM cluster and a gateway. The ISDN protocols, Q.921 and Q.931, are used only on an ISDN link to the PSTN, as illustrated in Figure 1-5.FXS 12 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide PSTN V SIP Q.921 Q.931 Figure 1-5 SIP Signaling SCCP SCCP works in a client/server architecture, as shown in Figure 1-6, which simplifies the configuration of SCCP devices such as Cisco IP Phones and Cisco ATA 180 Series and VG200 Series FXS gateways. PSTN V SCCP SCCP Endpoint Figure 1-6 SCCP Signaling SCCP is also used on Cisco VG224 and VG248 Analog Phone Gateways, in addition to analog telephone adapters (ATA). ATAs enable communications between Cisco UCM and a gateway. The gateway then uses standard analog signaling to an analog device connect- ed to the ATA’s foreign exchange station (FXS) port. Recent versions of Cisco IOS voice gateways—for example, the 2900 series—also support SCCP controlled Foreign Exchange Station (FXS) ports. Gateway Deployment Example Gateways are deployed usually as edge devices on a network. Because gateways might interface with both the PSTN and a company WAN, they must have appropriate hardware and utilize an appropriate protocol for that network. Figure 1-7 represents a scenario where three types of gateways are deployed for VoIP and PSTN interconnections.Chapter 1: Introducing Voice Gateways 13 San Jose UCM Cluster Chicago IP WAN V V MGCP UCME H.323 SJ-GW CHI-GW PSTN Denver SIP DNV-GW V IP SIP Proxy Server Figure 1-7 Gateway Deployment Example The scenario shown in Figure 1-7 displays the unified communications network of a com- pany that was recently formed as a result of a merger of three individual companies. In the past, each company had its own strategy in terms of how it connected to the PSTN: ■ The San Jose location used a Cisco UCM environment with an MGCP-controlled uni- fied communications gateway to connect to the PSTN. ■ The Chicago location used a Cisco UCM Express environment with an H.323-based unified communications gateway to connect to the PSTN. ■ The Denver location used a Cisco SIP proxy server and SIP IP phones as well as a SIP-based unified communications gateway to connect to the PSTN. Because the Denver location is only a small office, it does not use the WAN for IP telephony traf- fic to the other locations. Therefore, Denver’s local VoIP network is connected only to the PSTN. IP Telephony Deployment Models Each IP telephony deployment model differs in the type of traffic that is carried over the WAN, the location of the call-processing agent, and the size of the deployment. Cisco IP telephony supports these deployment models: ■ Single site ■ Multisite with centralized call processing14 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide ■ Multisite with distributed call processing ■ Clustering over the IP WAN Single-Site Deployment The single-site model for Cisco Unified Communications consists of a call-processing agent cluster located at a single site, or campus, with no telephony services provided over an IP WAN. Figure 1-8 illustrates a typical single-site deployment. All Cisco UCM servers, applications, and DSP resources are located in the same physical location. You can implement multiple clusters inside a LAN or a metropolitan-area network (MAN) and connect them through intercluster trunks if you need to deploy more IP phones in a single-site configuration. Cisco UCM Cluster PSTN V SIP/SCCP WAN Data Only Figure 1-8 Single-Site Deployment An enterprise typically deploys the single-site model over a LAN or MAN, which carries the voice traffic within the site. Gateway trunks that connect directly to the PSTN handle all external calls. If an IP WAN exists between sites, it is used to carry data traffic only; no telephony services are provided over the WAN. Design Characteristics of Single-Site Deployment The single-site model has the following design characteristics: ■ Single Cisco UCM cluster. ■ Maximum of 30,000 SCCP or SIP IP phones or SCCP video endpoints per cluster. ■ Maximum of 2100 H.323 devices (gateways, multipoint control units MCUs, trunks, and clients) or MGCP gateways per UCM cluster. ■ PSTN for all calls outside the site.Chapter 1: Introducing Voice Gateways 15 ■ DSP resources for conferencing, transcoding, and media termination point (MTP) services. ■ Voice-mail, unified messaging, Cisco Unified Presence, audio, and video components. ■ Capability to integrate with legacy PBX and voice-mail systems. ■ H.323 clients, MCUs, and H.323/H.320 gateways that require a gatekeeper to place calls must register with a Cisco IOS Gatekeeper (Cisco IOS Release 12.3(8)T or greater). UCM then uses an H.323 trunk to integrate with a gatekeeper and provide call-routing and bandwidth-management services for H.323 devices registered to it. Multiple Cisco IOS Gatekeepers might be used to provide redundancy. ■ MCU resources are required for multipoint video conferencing. Depending on con- ferencing requirements, these resources might be either SCCP or H.323, or both. ■ H.323/H.320 video gateways are needed to communicate with H.320 videoconfer- encing devices on a public ISDN network. ■ High-bandwidth audio (for example, G.711, G.722, or Cisco Wideband Audio) between devices within the site. ■ High-bandwidth video (for example, 384 kbps or greater) between devices within the site. The Cisco Unified Video Advantage Wideband Codec, operating at 7 Mbps, is also supported. Benefits of Single-Site Deployment A single infrastructure for a converged network solution provides significant cost benefits and enables Cisco Unified Communications to take advantage of many IP-based applica- tions in an enterprise. Single-site deployment also allows each site to be completely self- contained. There is no dependency for service in the event of an IP WAN failure or insuf- ficient bandwidth, and there is no loss of call-processing service or functionality. The main benefits of the single-site model are the following: ■ Ease of deployment. ■ A common infrastructure for a converged solution. ■ Simplified dial plan. ■ No transcoding resources are required because of the use of a single high-band- width codec. Design Guidelines for Single-Site Deployment Single-site deployment is a subset of the distributed and centralized call-processing model. Future scalability requires that you adhere to the recommended best practices specific to the distributed and centralized call-processing model. When you develop a stable, single site that is based on a common infrastructure philosophy, you can easily expand the IP telephony system applications, such as video streaming and videoconfer- encing, to remote sites.16 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide Follow these guidelines and best practices when implementing the single-site model: ■ Provide a highly available, fault-tolerant infrastructure based on a common infrastruc- ture philosophy. A sound infrastructure is essential for easier migration to Cisco Unified Communications, integration with applications such as video streaming and video conferencing, and expansion of your Cisco Unified Communications deploy- ment across the WAN or to multiple UCM clusters. ■ Know the calling patterns for your enterprise. Use the single-site model if most of the calls from your enterprise are within the same site or to PSTN users outside your enterprise. ■ Use G.711 codecs for all endpoints. This practice eliminates the consumption of DSP resources for transcoding, and those resources can be allocated to other functions, such as conferencing and MTPs. ■ Use SIP, SRST, and MGCP gateways for the PSTN. This practice simplifies dial plan configuration. H.323 might be required to support specific functionality, such as sup- port for SS7 or Nonfacility Associated Signaling (NFAS), which allows a single chan- nel on one digital circuit to carry signaling information for multiple digital circuits. ■ Implement the recommended network infrastructure for high availability, connectiv- ity options for phones (in-line power), QoS mechanisms, and security. Multisite WAN with Centralized Call-Processing Deployment The model for a multisite WAN deployment with centralized call processing consists of a single call-processing agent cluster that provides services for multiple remote sites and uses the IP WAN to transport Cisco Unified Communications traffic between sites. The IP WAN also carries call control signaling between central and remote sites. Figure 1-9 illustrates a typical centralized call-processing deployment, with a UCM cluster as the call-processing agent at the central site and an IP WAN with QoS enabled to connect all the sites. The remote sites rely on the centralized UCM cluster to handle their call pro- cessing. Applications such as voice-mail and interactive voice response (IVR) systems are typically centralized as well to reduce the overall costs of administration and maintenance. WAN connectivity options include the f ollowing: ■ Leased lines ■ Frame Relay ■ ATM ■ ATM and Frame Relay Service Inter-Working (SIW) ■ Multiprotocol Label Switching (MPLS) VPN ■ Voice- and Video-Enabled IP Security Protocol (IPsec) VPN (V3PN) Routers that reside at WAN edges require QoS mechanisms, such as priority queuing and traffic shaping, to protect voice traffic from data traffic across the WAN, where bandwidthChapter 1: Introducing Voice Gateways 17 is typically scarce. In addition, a call admission control scheme is needed to avoid over- subscribing the WAN links with voice traffic and deteriorating the quality of established calls. For centralized call-processing deployments, the locations construct within UCM provides call admission control. Cisco UCM Cluster V SIP/SCCP IP PSTN WAN SRST SRST Capable Capable V V SIP/SCCP SIP/SCCP Figure 1-9 Multisite WAN with Centralized Call Processing A variety of Cisco gateways can provide remote sites with PSTN access. When the IP WAN is down, or if all the available bandwidth on the IP WAN has been consumed, users at remote sites can dial a PSTN access code and place their calls through the PSTN. The Cisco Unified SRST feature, available for both SCCP and SIP phones, provides call pro- cessing at the branch offices for Cisco IP Phones if they lose their connection to the remote primary, secondary, or tertiary UCM server or if the WAN connection is down. Cisco Unified SRST functionality is available on Cisco IOS gateways running the SRST feature or on Cisco Unified Communications Manager Express (Unified CME) Release 4.0 and later running in SRST mode. Unified CME running in SRST mode provides more features for the phones than SRST on a Cisco IOS gateway. www.allitebooks.com18 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide Design Characteristics of Multisite WAN with Centralized Call-Processing Deployment The multisite model with centralized call processing has the following design characteristics: ■ Single UCM cluster. ■ Maximum of 30,000 SCCP or SIP IP phones or SCCP video endpoints per cluster. ■ Maximum of 1000 locations per UCM cluster. ■ Maximum of 2100 H.323 devices (gateways, MCUs, trunks, and clients) or 1100 MGCP gateways per UCM cluster. ■ PSTN for all external calls. ■ DSP resources for conferencing, transcoding, and MTP. ■ Voice-mail, unified messaging, Cisco Unified Presence, audio, and video components. ■ Capability to integrate with legacy PBX and voice-mail systems. ■ H.323 clients, MCUs, and H.323/H.320 gateways that require a gatekeeper to place calls must register with a Cisco IOS Gatekeeper (Cisco IOS Release 12.3(8)T or later). UCM then uses an H.323 trunk to integrate with the gatekeeper and provide call- routing and bandwidth-management services for the H.323 devices registered to it. Multiple Cisco IOS Gatekeepers might be used to provide redundancy. ■ MCU resources are required for multipoint video conferencing. Depending on con- ferencing requirements, these resources might be either SCCP or H.323, or both, and might all be located at a central site or might be distributed to the remote sites if local conferencing resources are required. ■ H.323/H.320 video gateways are needed to communicate with H.320 videoconfer- encing devices on a public ISDN network. These gateways might all be located at the central site or distributed to the remote sites if local ISDN access is required. ■ High-bandwidth audio (for example, G.711, G.722, or Cisco Wideband Audio) between devices in the same site and low-bandwidth audio (for example, G.729 or G.728) between devices in different sites. ■ High-bandwidth video (for example, 384 kbps or greater) between devices in the same site and low-bandwidth video (for example, 128 kbps) between devices at dif- ferent sites. The Cisco Unified Video Advantage Wideband Codec, operating at 7 Mbps, is recommended only for calls between devices at the same site. ■ Minimum of 768 kbps or greater WAN link speeds. Video is not recommended on WAN connections that operate at speeds lower than 768 kbps. ■ UCM locations provide call admission control, and automated alternate routing (AAR) is also supported for video calls, which allows calls to flow over the PSTN if a call across the WAN is rejected by the locations feature.Chapter 1: Introducing Voice Gateways 19 ■ SRST versions 4.0 and later support video. However, versions of SRST prior to 4.0 do not support video, and SCCP video endpoints located at remote sites become audio- only devices if the WAN connection fails. ■ Cisco Unified CME versions 4.0 and later might be used for remote site survivability instead of an SRST router. Unified CME also provides more features than the SRST router during WAN outage. ■ Cisco Unified CME can be integrated with Cisco Unity Express (CUE) in the branch office or remote site. The Cisco Unity server is registered to the UCM at the central site in normal mode and can fall back to Unified CME in SRST mode when the central- ized UCM server is not reachable, or during a WAN outage, to provide the users at the branch offices with access to their voice mail with message waiting indicators (MWI). Design Guidelines for Multisite WAN with Centralized Call-Processing Deployment Follow these guidelines when implementing the multisite WAN model with centralized call processing: ■ Minimize delay between Cisco UCM and remote locations to reduce voice cut- through delays (also known as clipping). The ITU-T G.114 recommendation specifies a 150 ms maximum one way. ■ Use HSRP for network resiliency. ■ Use the locations mechanism in Cisco UCM to provide call admission control into and out of remote branches. ■ The number of IP phones and line appearances supported in SRST mode at each remote site depends on the branch router platform, the amount of memory installed, and the Cisco IOS release. SRST on a Cisco IOS gateway supports as many as 1500 phones, whereas Unified CME running in SRST mode supports 240 phones. Generally speaking, however, the choice of whether to adopt a centralized call- processing approach or distributed call-processing approach for a given site depends on a number of factors, such as ■ IP WAN bandwidth or delay limitations ■ Criticality of the voice network ■ Feature set needs ■ Scalability ■ Ease of management ■ Cost Note If a distributed call-processing model is deemed more suitable for a customer’s business needs, the choices include installing a UCM cluster at each site or running Unified CME at the remote sites.20 Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation Learning Guide ■ At the remote sites, use the following features to ensure call-processing survivability in the event of a WAN failure: ■ For SCCP phones, use SRST on a Cisco IOS gateway or Unified CME running in SRST mode. ■ For SIP phones, use SIP SRST. ■ For devices attached to analog or digital voice ports, use MGCP Gateway Fallback. SRST or Unified CME in SRST mode, SIP SRST, and MGCP Gateway Fallback can reside with each other on the same Cisco IOS gateway. For specific sizing recommendations, refer to the Cisco Unified Communications System SRND based on Cisco UCM 8.x at the following link: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/uc8x.html. Multisite WAN with Distributed Call-Processing Deployment The model for a multisite WAN deployment with distributed call processing, as illustrated in Figure 1-10, consists of multiple independent sites, each with its own call-processing agent cluster connected to an IP WAN that carries voice traffic between the distributed sites. An IP WAN interconnects all the distributed call-processing sites. Typically, the PSTN serves as a backup connection between the sites in case the IP WAN connection fails or does not have any available bandwidth. A site connected only through the PSTN is a standalone site and is not covered by the distributed call-processing model. WAN connectivity options include the f ollowing: ■ Leased lines ■ Frame Relay ■ ATM ■ ATM and Frame Relay SIW ■ MPLS VPN ■ IPsec V3PN Multisite distributed call processing allows each site to be completely self-contained. In the event of an IP WAN failure or insufficient bandwidth, a site does not lose call-pro- cessing service or functionality. Cisco UCM simply sends all calls between the sites across the PSTN.

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