Dave SchulzSecond Edition Dave Schulz ciscopress.com bee ec Cra a) Perec CaP Cat ee ene) Pere ee Cray Per eaten eS eraTet Crary ron eet LR areD eee Cray Call Control Discovery .(21 GONP Voice GIPT2 642-457 Quick Reference, Second Edition About the Author Dave Schulz has more than 25 years of experience with various technologies, ranging from routing and switching to security and voice technologies, Before joining Skyline Advanced Technology Services, he was involved in network engineering and consulting, project management, and oversight of engineering and maintenance activites for a reseller in the Midwest. He has also taught various technologies to customers and engineers, and created various process and procedure methodologies, service pricing, and documentation. Dave ereated a Technical Assistance Center, while being a manager and director of professional services and performing installation, support, and consulting in various customer environments, He also hae contracting responsibilities ata large, global enterprise level corporation, where duties varied from routing, switching, security, wireless, and project management, Dave currently teaches voice technology classes throughout the United States for Skyline Advanced Technology Services and is involved in course de-velopment. Dave resides in Cincinnati, Ohio, with his wife, Peggy, and three daughters, Amy, Ericka, and Tiffany. About the Technical Editor Alex Hannah, CCIE Voice No. 25853, isa certified Cisco instructor, specializing in teaching the Cisco Advanced IP Communications product line, He has over 7 years consulting experience in Cisco Unified Communications for ‘SMB through Enterprise spaces, He is president of Hannah Technologies LIC, a Richmond, Virginia based Cisco ‘consulting firm specializing in Ciseo Advanced IP Communications and application development using Microsoft technologies, He holds a bachelor's degree in Information System from Virginia Commonwealth University with ‘a minor in Business, Additionally he is the founder of UCCX.net a video based training website for the Cisco UC product line, In his spare time, you can find Alex on his boat wakeboarding with his family and friends 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta13] GONP Voice GIPT2 642-457 Quick Reference, Second Edition Chapter 1 Multisite Deployments ‘This chapter reviews issues and solutions related to mulisite deployments It also covers the connections between these sites and d plans for such configurations Issues ‘This section identifies the issues that can arse ina multisite Cisco Unified Communications Manager (CUCM) deployment, Issues Overview Following are the major issues that can rest se problems fr voice packets '© Bandwidth: Optimizing the use of bandwidth for voice and data, which compete for bandwidth '© Availabilty: Addressing fallback solutions that acd tobe implemented for critical services by providing backup paths for trunks and gateways '© Dial plan: Configuring the dal plan for optimum routing considering overlapping directory numbers, E164 dialing, DIDs, tail-end hop-off, ol bypass, and PSTN backup © NAT and Security: Connecting tothe Internet using translation resulsin visi can be used between sites 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta141 Ghaptert Mutsite Deployments Solutions ‘This section reviews the solutions to issues that occur in Cisco Unified Communications Manager (CUCM) multisite deployments Quality of Service Queuing in biter am cause tail drop which esas in packet drops The same queuing ses cam also delay, reskin in jer ter isthe variable delay of recived packet Both ely and iter affect th oie quay. To sl the delay and jter iss you can use ual of Serce (QoS), After fis identified and dived int clases, you can apply a QoS palicy perl. With QoS enabled, toic afi is proited over al other trafic because itis lnteneysensive tafe, Voice afi is UDP and tansmited using Real Time Transp Protocol (RTP). Lost packets ae no reransmted, Only RTP sequence numbers cn ensure ta packets ive a thir destination inthe proper oder. Therefore, queuing, delay, and dropouts tend to havea negative impact on voce ac. Bandwidth {Ina multisite envizonment, voice, data, and video compete forthe same bandwidth on the WAN. QoS can prioritize voice trafic However, bandwidth i limited and must be conserved properly using the following deployment options 1 Low-bandwvidth codecs, such as G.729 across the WAN 1 RTP-header compression for low-speed WANs (under 768 k) Following are media resource considerations 1B Use the local media resources. Those media resources canbe both software and hardware resources, depending onthe needs However, the selection of which resource to use is governed by the media resource groups (MEG) and media resource sroup (MRGL) configuration. You need to primarily use resources at the location where the nced originates and use the remote resources asa backup or secondary choice. 18 Deploy local and mixed conference bridges, Media Termination Points (MTP), transcode, or conference bridges, 1B Deploy local Music on Hold (MOH) servers, and use multicast MOH from branch router flash 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta15] Ghaptert Mutsite Deployments Consider signaling, media transmission, and service across the WAN in a centralized deployment, which results in availability issues when the WAN is down, Provide redundancy, PSTN backup, and Surivable Remote Site Telephony (SRS) In amultste deployment you need to consider a number of issues abou the dial pln issues, Provide optimized call routing that enables various numbering plans (fixed or variable), tll bypass, til-end hop-off, and PSTN backup to ensure availabil- ity. Fived numbering plans should be the goal of any new implementation; however, this might not be the reality considering the current business environment, acquisitions, or other elements affecting the organization. and provide redundancy and backup paths ‘The dial plan configuration must employ opt for remote locations, Scalability ofthe us use features, such as extension mobili Limit dhe numberof voice cal admission control (CAC) and provide backup using Automatic Altemate Routing (AAR). Deploy locations, RSVP, SIP precondition, or Gatekeepers to provide CAC. Consider vulnerability om the Internet in a multisite deployment, when providing acess to remote service ofthe ITSP. Provide NAT translation, VPN tunnels for remote locations, ad Cisco Unified Border Element (CUBE) conigured for ITSP connectivity, Availability Availity issues ae areas by providing the following PSTN backup MCP flock Fallback for IP Phones Call Forward Unregistered (CFUR) 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta[0] Ghaptert Mutsite Deployments Dial Plan Solutions include the following: Access codes and s codes for interste dialing of multisite deployments, ‘Some countries have variable-length numbering plans tha use the overlap sending and receiving feature. Use of overlap sending and receiving and # for variable-length numbering plans needs to be considered. Overlap sending and receiving is ‘more prevalent in countries ouside ofthe United States. Use of IVR for remote auto attendant and direct inward dialing (DID) ranges and E164 addressing. '© Different number presentation in ISDN (Type of Number [TON], whichis also referred to asthe Type and Plan forthe calling '© For international multisite deployments, deploy a globalized ds and called number) ‘Toll bypass, tail-end hop-off(TEHO), and PST backup to provide las cost routing for prioritize and optimized call Dial plan considerations include endpoint addressing, call outing and path selection, digit manipulation, calling privilege, and call cover lan using the E.164 dialing with + dialing capabilites, 1A plobalized il plan provides called and calling numbers using E168 format with the + prefix, After th calls routing, the ‘number can be provided in a globalized or localized formal, Localized format is defined as dialing according the local dial ll routing can be viewed in tree phases, from ingress to routing (according the dil plan) and finally ingress must be normalized, or globalized, before call outing The globalized number (matching E164 format) ‘must be localized at egress. The globalized format provides these advantage: ‘One format required for call routing E,164 format) Speed dials, fase dials, and features available when using mobility ‘Common format for global directories ‘Simplified dial plans for global newworks 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta71 Ghaptert Mutsite Deployments Dial Plan Scalability Issues Dial plans can become complex and dificult to implement and manage in large, enterprise networks, The issues fallow: 1 Static configuration is vey complex and dificult to manage 1 Centralized 41323 gaceepers simpli this configuration but provide no dynamic recognition and automatic PSTN rerouting © Call Control Discovery (CCD) enables dynamic learning of dial plans by dynamically exchanging call-routing information between clusters in Cisco Unilied CM. In this way, the any-lo-any static configuration of dil-pattem is no longer required NAT ‘To solve NAT security issues, use CUBE as an application proxy. This canbe deployed in two scenarios: 1 Signating Only (low around) 1 Signaling and media ow through) ‘The flow-through method provides a higher level of security for hiding inside addresses. RSVP, SIP precondition, trusted relay points, and proxies inthe ASA provide additional options. Hardware Digital Signal Processor (DSP) resources are required and can significantly add to the cost ofthe implementation. Software MTPs can be used only when configuring pass-through. Multisite Connections ‘This section reviews the configuration of gateways and trunks ina follows: i cnvironments where the four deployment issues areas = Availabi y issues: Solved by SRST and MGCP fallback © Quality and bandwidth issues: Slved by QoS, CAC, RTP header compression, and local media resources 1 Dial plan issues: Soived by the use of access and site codes and digit manipulation '= NAT issues: Solved by the deployment of CUBE 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta18] Ghaptert Mutsite Deployments MGCP Gateway Implementation First, add the MGCP gateway to Cisco Unified Communications Manager. Then, add MGCP endpoints tothe gateway, and configure the endpoints. Next, configure the gateway, CUCM stores an XML configuration flim ts TFTP server. Us the fllowing commands 1 com-manager config server 1 ccm-manages contig “The later command initiates the download ofthe XML. configuration from the TFTP server configured using the cem-manager config server command. These commands can safely be removed afer the configuration file has been downloaded but must be re-entered when changes are made tothe configuration file for the gateway in CUCM. H.323 Gateway Implementation ‘The Cisco 10S gateway configuration for H373 includes the following tps Step 1. Configure the H.323 gateway, specifying its H323 ID andthe IPadress Table 4-1 Digit Manipulation Requirement Step 2. Configare one of more VP dil pers pointing to CUCM. Step 3. Configure one or more POTS dil pers pointing othe PSTN. Dia pers provide the cal routing for H.323 enews Following isa sample configuration for H.323: 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta191 Ghaptert Mutsite Deployments ho2igatevay voip bind sresdae 10.1.1.50 dszect-snvard-dial pot 07/0/0123 Inthe preceding configuration, the voice service voip command isnot required for H,23 because H.323 is the default appli tion protocol used by the 1OS-based gateway. However, the h22S timeout command is required for call preservation during switchback for MGCP fallback. MGCP fallback is explored inthe next section, ‘The codec command forthe frst dial-per defines the codec used for communication to CUCM (g.7Llutaw), Ifeodee negotia- tion is required between multiple codec, use the voiee class codec command, The configuration of this voice cassis explored in the Implementing Cisco Voice Communications and QoS (CVOICE) course Implementing SIP Trunks To a Sesion Iniaton Protocol (SI) trunk in CUCM, complet the following steps: ‘Stop 1. Navigate to Device> Trunk and click Add New, ‘Step 2. Inthe Trunk Type drop-down list, choose SUP Trumk and cick Next, ‘Step 3. Inthe Trunk Configuration window, enter a name and description forthe SIP trunk, and select the device poo! 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta1101 Ghaptert Mutsite Deployments ‘Step 4. Inthe SIP Information area ofthe Trunk Configuration window, enter the destination address Step. Selecta SIP Trunk Securiy Profile anda SIP Profile Route Group and Route List Configuration JnCUCM, you need to complete the fllowing configuration fo all satevay selection, Complete the following Stop 1. Create the gata’ and spi its IP adres, Step 2. Create a Route Group and athe gateway crested in Step 1 Step 3. Create a Route List and add he Route Group crested in Sep 2 Step 4. Crete one of more Rout Paitems pointing tthe Rou Ls. ways. The use of route groups and route lists determines the Dial Plans for Multisite Environments ‘This section reviews the implementation of a dial plan to support inbound and outbound PSTN dialing, ste-sode dialing, and tai-end hop-of Overview Dial plan solutions fora multisite environment with centralized or distributed call processing include the following 1B Access and site codes © mplementing PSTN access © Implementing PSTN backup Globalized Dial Plan using E164 forms 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate detain Ghaptert Mutsite Deployments Implementing Site Codes for On-Net Calls Site-code dating is used when two sites have overlapping and nonconsccutive directory’ numbers, For example, uses ial an access code followed bya three-digit ste code, When distributed call processing is used, cach CUCM cluster is only aware ofits own directory numbers, For al directory numbers located a the other st, the eal is outed to a CUCM server a the othr site based on the dialed site code. You must remove the acces nd site codes fom the Dialed Numbor Identification Service (DNIS) fr outgoing callocasure proper cll outing. The sit cades can be any number f digits but should be unique tothe current dia plan For incoming calls, the access code an site code forthe CallerID of the caller must be added: this procedure is done using translation patterns, "This provides the proper information required for callback. overlapping directory numbers exit in a centazed cll-processing deployment, you must implement acess and ste codes ina dferent way: In this case, you need to deploy partitions and calling search spaces (CSS) so that phones at the remote site do notsee dretory numbers of mainsite phones and vce Versa, In hese cases, dela translation pti per sie to provide prope call routing functionality For example, Site A and Site B both have extensions 1001 to 1999, configured withthe search spaces and partitions for their site Ifusers at Site dial any extension in this range, they will be directed to a phone at their local site ‘To provide access to the remote ste, an access code was selected: 85 was selected for Site A and 86 was selected forthe ste code for users at Site B to access users on Site A. In this ease, when users needed to calla user at Site B, dhey imply dial 88, followed by the four-digit extension ofthe user. translation patie was configured with the partition configured in the search space for the Site A users, The translation pater performed the following functions: 1B Matched digit analysis om the 85, XXX pattern, 1 Performed Pre-Dot discard digit instruction (DDN, by removing the 85 from the dialed string. 1 The translation pattern used the calling search space of Site B, which has acess tothe extensionparitions at ite B. Additionally, the calling numbers changed to ad the site code for the Site A will ow witere the call originated and can return the call from the various '© Similar configuration neds to be completed for users at Site B to call users at Site A ion, In this way, the called party at Site B 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta121 Ghaptert Mutsite Deployments With this configuration, a user at Site A with extension 1082 places a call to a user ut Site B with extension 1075, To place the 1451075, The dialed number matches the translation patter in which the 85 is stripped and calling search space is changed tothe search space configured at Site B. Digit analysis is again performed on the remaining digits of 1075, ‘which now has access tothe partitions of Site B. The inal results thatthe phone with extension 1075 rings at Site B. This phone also displays the calling number of $1052, informing the called party that the call was placed fram Site A, Implementing PSTN Access ‘When implementing PSTN access, both outgoing transformation and DNIS transformation, and incoming calls require Automatic Number Identification (AND) For outgoing calls, ensure the following 18 Po dicot inward dialing range (DID) i used at the PSTN, transform all directory numbers to the same, single PSTN. number inthe ANL = IFDID is used, extend the directory numbers toa full PSEN number. 18 For the DNIS transformation, strip the access code o provide proper routing, '© For incoming calls, transform ANI o the full number, and add the proper site or access code of the Cal callback information, © IF DID is used strip off the office code, aca code, and country code to provide proper call routing of the directory number ID to provide proper © IF DID isnot used, route the incoming calls to an attendant or interactive voice response (IVR) application. Implementing Selective PSTN Breakout Must deployments fen feature one PSTN gateway per sit. Selective PSTN breakont ensures tha sal gata am acess the PSTN, Thisfnctionali is provided trough the configuration of th varioas Route Pats. These Route Pats ae pa ito iret prions, which point o dierent PSTN gateways in the network 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta1131 Ghaptert Mutsite Deployments Finally, the IP Phones need to be configured with a CSS that ineludes only the Route Patterns that refer to their local gateway ‘This functionality can also be configured using the Local Route Group, which defines the gateway configuration within the de- vice pool of the IP phones, Considerations When Using Backup PSTN Gateways ‘You have two options for configuring the ANI forthe outgoing cll. Use the primary gateway's PSTN number on the secondary snteway or use the secondary gateways PSTN number, For example, when calls placed to a remote user using the PSTN breakout af the local galeway, the called party aces the galway number displayed on his phone and call listo return the call. The choice to use the scondary or primary gateway’ for he retum calls depends onthe organization, service provider, and gateway configuration. Not ‘You cannot configure an AN! that snot owned by the organization forthe selected carirand locaton. The service provider might ovennrite this information, with the configured circuits ANI Implementing PSTN Backup for On-Net Intersite Calls Intersitecals should use the intercluster trunk (ICT) over the IP WAN. Ifthe IP WAN is down, the PSTN should be sed as a backup for intetsite call, PSTN backup for on-net calls can be easily provided by Route Lists and Route Groups, giving priority tothe ‘intercluster trunk versus the PSTN gateway Implementing Tail-End Hop-Off ‘When implementing ‘TEHO, PSTN breakout occurs atthe gateway closes tothe dialed PSTN destination. Create a Route Patten for each destination area that can be reached at diferent coss, These Route Pattsns are required once per site and must be put into diferent partitions if not using a globalized dil plan. globalized dial plan is discussed in the next section, The globalized dial plan significantly reduces the number of route patterns required by using the global partion to provide access to a route list for access 10 specific gateways for TEHO, while using local route groups for backup. Considerations for Globalized Dial Plans ‘When the dial plan requires the use of different numbering plans, consider te implementation ofthe globalized dial plan using E164 format. When a user dials a number according tothe local dial ruts, the dial number is nommalized or globalized to E.164 format before routing tothe gateway, where thecal is then transformed tothe localized format, The + dialing is supported forthe translation pater, transformation masks, and Route Patter to be used forthe DNIS and ANIL E.164 dialing has a maximum of 15 digits 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta141 Ghaptert Mutsite Deployments and is prefixed withthe +, whic is designated asthe international E164 escape character. Thereby calls can be placed anywhere i the world using the same number. For ecample, 1 dal the number in San Jose of 14088550155, a user could dial the number, “+14085550155 from anywhere in the world using a configured globalized dial plan 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta115) GONP Voice GIPT2 642-457 Quick Reference, Second Edition Chapter 2 Centralized Call Processing Redundancy This chapter reviews the mechanisms for providing call survivability and device failover in remote sites. This includes the configuration of Cisco IOS routers as Suvivable Remote Site Telephony (SRST) gateways with MGCP fallback, and the use of Cisco Unified Communications Manager Express in SRST mode. Options This ection eviews the mechan ims for providing cll suvivabiliy and device fulover in emote sites, SRST and MGCP gate nay fallback ae the hey components to deliver fail-safe communication services. To use SRST as your fallback mode on an MGCP gateway, SRST and MGCP fallback must be configured on the same gateway. SIP SRST provides a basic set of features to $1P-based IP Phones, It has to be enabled and configured separately on Cisco 10S routers. In Cisco Unified SRST versions prior to 34, it provides a SIP Redirect Server function: in subsequent versions it ats as back-to-back user agent (22BUA), SRST uses SIP and H.323 dial-peer to provide backup when the CUCM is unavailable, regardless ofthe protocol used for communication to CLICM during normal operation, H.323 isthe most prevalent default protocol used for SRST. Note: ‘Additional licensing may be required for routers with IOS version 15. CLIC Express in SRST mode provides more features o a smaller maximum number of Skinny-based IP Phones by falling back to CUCM Express mode MGCP gateway fallback configured as an individual feature can be used by a PSTN gateway if H.323 or SIP is configured as the default application, meaning that one of these peer-to-peer protocols will be used when MGCP loses connectivity othe call agent However, as mentioned previously, H.325 isthe default protocol used by 1OS-based gateways, ered. Thi pubcatin potted by copyright Please oo116) Ghapter 2 Centralized Gall Processing Redundancy CCiseo Unified SRST enables routers to provide basic eall-handling support for Cisco Unified LP Phones when they lose eonnee- tion to remote primary; secondary, and tertiary CUCMS, as when the WAN connection is down in a centralized environment Cisco Unified SIP SRST provides backup to an extemal SIP proxy server by providing basic registrar and redirect server ser- vices or B2BUA services, CLICM Express in SRST mode enables routers to provide basic call-handling support for Cisco Unified 1P Phones if they lose connection to remote primary, secondary, and tertiary CUCMS, as when the WAN connection is down, Cisco Unified SRST Operation CUCM supports Cisco Unifid IP Phones at remote sites attached to Cisco multiservice routers across the WAN. The following details the operation ‘The remote-site IP Phones register with CUCM, Keepalive messages are exchanged betwcen IP Phones and the central CUCM across the WAN, ‘The CUCM atthe main ste handles the call processing forthe branch IP Phones. |When the WAN link fils, the IP Phones lose contact with the central CUCM but then register with the local Cisco Unified SRST gateway. 1% The Cisco Unified SRST gateway detects newly registered IP Phones and quties these IP Phones for thir configuration, which then autoconfigues itself 1% The Cisco Unifiod SRST gateway uses Simple Network-Enabled Auto Provision (SNAP) technology to autoconfigure the branch office router to provide call processing for Cisco Unified IP Phones registred with the route. 1 Cisco Uniti IP Phones attempt to reestablish a connection withthe CUCM at the main site periodically when they are regis- tered witha Cisco Unified SRST gateway, based on the keepalives sent to CUCM. typically takes three times the keepaive period for a phone to discover that its connection to CUCM has fa keepalive period is 30 seconds ld, The default 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta“7 Ghapter 2 Centralized Gall Processing Redundancy MGCP Fallback Operation [MGCP gateways register with the CUCM, MGCP gateways exchange kes withthe central CUCM across the WAN, CUCM isthe MGCP call agem. The MGCP gatnay performs a switchover tits default technology when the keealives between CUCM and the Cisco MGCP gateway are missing. The switchback or re-home mechanism is triggered by the restablshment ofthe ‘TCP connection between CUCM and the Cisco MGCP gateway Dial Plan Requirements for MGCP Fallback and SRST Scenarios ‘SRST failover means thatthe remote sit is independent from the complex dial plan implemented én CUCM at the main site, The ‘SRST router needs to havea minimal dal plan implemented, This is accomplished using voip and pots dil pecs. During fallback, users should dial main-site directory numbers as usual Because these calls must be routed over the PSTN, during fallback, the main-site extensions must be translated to E.164 PSTN numbers atthe PSTN gateway SRST use the local PSTN breakout, while supporting call preservation, autoprovisioning, and failover. To guarantee PSTN, connectivity, dial peers with destination pattems corresponding tothe PS'TN access code must be implemented in H.323 or SIP gateways. CLICM considers the remote-site phones as unregistered and cannot route calls tothe affected IP Phone directory num= bers. Therefore, if main-site users dial internal extensions during the IP WAN outage, the calls will fail, The Call Forward Unregistered (CFUR) configuration address this issue, CFUR provides additional call forwarding capability when the phone becomes unregistered for any reason. Cisco Unified SRST 8.0 is supported in IOS gateways and CUCM Express version 15,0(1), SRST8.0 supports eight calls per line, E.164 numbering (using the + prefix), and five addtional MOH stream when using SCCP. The number of phones supported in SRST'8.0 depends on the platform, with a maximum of 1500 phones supported en the 3945 platform. ‘Tallow remote IP Phones to be reached from main-site IP Phones, CFUR must be configured forthe remote-site phones. Ifa slobalized dial plan i configured, CFUR can provides access to calling via a single Route Patter (+!) and CSS forall SRST calls. These ealls are placed using the desired gateway via the PSTN; however, you an also use CFUR to direct the caller toa local extension, operator, or voice mail 122011 Cisco Sytem ne lights reared. The publction protected by copyright Place soe page 0 for are detare) Ghapter 2 Centralized Gall Processing Redundancy When using globalized eall routing with Local Route Groups, the CFUR CSS is the same forall phones, so the local gateway {is used for these calls. n the case of remote locations, you do not need to use the IP WAN to the main site and breakout to the PSTN for CFUR calls to other remote sites. The CFUR eall uses the local gateway, resulting in an impraved call routing. Under normal conditions in mulisite deployments with centralized call processing, calling privileges are implemented using partitions and calling search spaces (CSS) within CUCM. When IP WAN connectivity is lost between a branch ste and the cen- tral site, Cisco Unificd SRST takes contral ofthe branch IP Phones, and the entire configuration related to partitions and CSSs is unavailable. Therefore, when running in SRST mode, the various lasses of service within the branch router must be defined in the SRST router using the clas of restriction (COR) functionality Cisco Unified SRST and MGCP Fallback ‘This ssction reviews the configuration of SRST to provide call survivability and MOH for SCCP phones and MGCP fallback for gateway survivability Overview Activate and configure the MGCP-gatoway-fallback feature on the Cisco 10S router, Configure Cisco Unified SRST on the side of the CUCM and onthe side ofthe Cisco 10S router. Configuration Follow thes tps for configuration: Step 1. Define the SRST references for phones an the CUCM. The default TCP pot is 2000 for SCP and 560 for SI ‘Step 2. MGCP fallback and Cisco Unified SRST are enabled and configured on the IOS gateway; use the call: manager fll- back command, 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta119) Ghapter 2 Centralized Gall Processing Redundancy Following i a sample configuration: l¢ configure terminal fi (config}# eadl-manager-fal.back Following is a sample MGCP- 1way-fllback configuration for an SRST:enabled Cisco 10S router: igure termina Al(contig)# commanager faltback-mgcp Al(conig)# application i (config-app)# global fl(contig-app-giobal)# service alternate Default 11 (config-app-giobal )# nd ‘Stop 3. Use an additonal command for call preservation when using MGCP fallback votes service voip ‘Stop 4. Configure the CFUR feature on the CUCM to reach remote sites in SRST mode, Also, configure the Max Forward Unregistered Hops to DN option inthe Service Parameters A reasonable setting of 2 for this option can eliminate routing loops, Step 5. Implement a simplified dial plan on the remole-site gateways to provide prope calling privileges and dialing behavior by using dial-peers and voice translation profiles. 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta120] Ghapter 2 Centralized Gall Processing Redundancy Cisco Unified Communications Manager Express in SRST Mode This section reviews the configuration of CUCM Express to provide telephony service, basic hunting, and Music an Hold (MOH) to ‘Skinny Client Control Protocol (SCCP) and session initiation protocol (SIP) phones ifthe connection tothe centralized call agen is lost, You can use CUCM Express in standalone mode in normal operation for small to medium sites (up to 450 phones). However, using CUCM Express provides more features to remote users when in SRST mode than standard Cisco Unified SRST. Providing Phone Loads 1 Phone Firmware files are available through the TFTP server. The ttp-server flash:-filename command enables the specified file that resides in flash memory to be downloaded via TFTP. New Features Supported in SRST 8.0 CUCM Express SRST mode now suppor the folowing fests: Sis music on-old sources can be conigred Support for E164 numbers and + proixes Use ofthe dial pattem command enbancements profile enbancements Voice transla 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate detai211 Ghapter 2 Centralized Gall Processing Redundancy Music on Hold Five additional music on-hold sources are supported in SRST 80 usingau and way file formats, (Six music on-hold source files are supported.) Phone configuration is based on MOH groups; otherwise, the default MOH source is used. Music on hold is enabled with the moh command under the tlephony service, Use the multicast moh command for multicast MOH functionality. Only G.711 is supported. Transcoders ar required for G.729 support Configuring Cisco Unified Communications Manager Express in SRST Mode ‘An SRST reference in CUCM can be a standard SRST gateway or a CUCM Express router. No phones need to be configured because thy can be learned by Simple Network-Enabled Auto Provision (SNAP). You also can do the following ‘© Manually configure ephones with associated ephonc ns "© Manually configure ephones with no associated ephone-da (auto assignment) "= Manually configure ephonc-dns (-phone configuration Ieamed by SNAP), Complete SRST provisioning Perform the configuration of CUCM Express in SRST mode in lelephony-service configuration mode. When the telephony- service command is active the eall-manager-fallback command is not accepted by the CLI, and vice versa To enable SRST mode for CUCM Express, use the srst mode auto-provision command in telephony-service configuration mode ‘The following sample configuration determines that both the ephones and das will be leamed by auto-provision using the dn template I fr the ds and ephone template 3 forthe ephones. These learned configurations are written to the show running contig ‘his is determined by the configuration forthe srst mode autoprovision command. Ifthe keyword “none” is configured here, no configuration is written to the show running-configuration, ‘telephony-service 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta122] Ghapter 2 Centralized Gall Processing Redundancy axtophone 15 erat ephore description aret auto-prov phone ‘The following configuration information illustrates the ephones and ephone-dns learned using the specified template. You also notice tha the description is applied from the tlephony-service configuration, ephone-tanplate 3 123432341296 ephone-tanplate 3 seo-prow phone fephone da 1 dual-line cphone-dn-template 1 cphone-dn-template 1 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta123] Ghapter 2 Centralized Gall Processing Redundancy ‘The following configuration provides an example of the ephone-dn and ephone template used with SRS'T fallback as applied to the telephony-service configuration: pickup-aroup 12 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta124) GONP Voice GIPT2 642-457 Quick Reference, Second Edition Chapter 3 Bandwidth Management and Call Admission Control This chapter reviews methods of reducing allocating bandwidth and using Call Admission Control (CAC). Bandwidth Management This section reviews bandwidth conservation techniques for remote locations across the WAN, Codec Configuration Use high-bandwidth codecs in LAN enionmens and use loy-bandvidth codes in WANs, Use an allemative method than a low tanith code for Mol These methods includ using mls fom the ranch office router or dialing Mal acrosthe WAN. The code usd depends onthe repion configuration in CUCM. Local Conference Bridge This keps tai off the IP WAN, Ths implemented using DSPs atthe routers ofthe remote sites, Design consideration and placement of conference bridges depend onthe amount of resources using he conference fete availble bandwidh, and cot of DsPs (22011 Cisco Systeme Ie lights reserved. Thi pulbicetn protected by copytight lasso128) Chapter 3 Bandwidth Management and Gall Admission Control, Transcoders ‘These media resources ae implemented in hardware enabling low-bandwidth codecs to be used if low-bandidth codecs are not supported by both endpoints. Transcoders provide the conversion mechanism between diferent codecs. ‘To configure transcoders, perform the following steps Step 1. Add the wanscoder resource to CUCM using Media Resource > Conference Bridge > Add New. ‘Step 2. Configure the tanscoder resource in Cisco 10S Software ‘Stop 3. Configure Media Resource Groups (MR) Step 4. Configure Media Resource Group Lists (MRGL). Step 5. Assign MRGLs to devices. Multicast MOH from Branch Router Flash ‘The CUCM must be configured for multicast MOH, andthe Cisco 10S router requires features tha support SRST. The locally generated MOH steam must be identical to the CUCM steam and must be G.711, Upto six Molt streams can be configured for ‘mubicat or unicast on a rouer using SRST 8 0, However, multicast must be used when using CUCM with CUCM Expres, by incrementing the various teams by address or ports. This configuration is complete inthe administration pages of CUCM. When MoH sources are configured, four files are ereated in G, 71 Lmu-law, G,711a-law, 6,729, and Wideband codees, ‘To configure multicast routing, perform the following steps: ‘Step 1. Enable multicast routing on the router. This is completed by adding the following commands to the existing configuration 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta128) Chapter 3 Bandwidth Management and Gall Admission Control, ip multicast-routing [Enable multicast MOH using Media Resources > Music On Hold Server, Stop 4. Configure the maximum number of hops ‘Step 5. Enhance the SRST configuration onthe router with the mo and multicast moh commands, ‘Step 6. — Multicast traffic can be limited by the following means 1B Use an access-list (ACL) to drop multicast packets atthe WAN. In this example, all multicast wai in the address range of | ‘using port 16384 willbe dropped, whereas all other waffic i a -Lint extended drop-mticart 1B Disable multicast on the WAN interface by removing the ip pim statement 1 Set the maximum hops command on the Music On Hold Server page to be drop traffic based on the expired TTL, 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta1271 Chapter 3 Bandwidth Management and Gall Admission Control, Implementing Call Admission Control ‘This section reviews call admission contol (CAC) in CUCM. CAC can prevent the oversubscription of voice calls based ona fixed (QoS-enabled bandwicthto the remote site, When oversubsription occurs, CAC can either deny the call or provide an altemative using Aulomated Altemate Routing (AAR), Overview CUCM supports CAC for centralized call processing using the following: Locations 1 RSVP-enabled locations. AAR provides an alternative o route calls over the PSTN during a CAC condition of oversubscripson, For distributed processing, the following is supported: '© 11323 gatckeeper or SIP precondition. "© Route Lists and Route Groups provide an alternative to route calls over the PSTN during oversubscription for distibuicd en Locations Calls re limited by permitting a certain bandwidth for all calls coming into or going out of a location. Locations are based on a hub and-spoke topology using a 20 mS packetization period (80 kbps for G.711, 24 kbps for G.729). To configure location perform the following: ‘Step 1. Configure the location using System > Location > Add New. Step 2. Assign locations to device pools (mandatory) and to devices using Device > Phone. Location configuration canbe ap- plied in both places: device pool and device level, The configuration at the device-level overrides the device pool con figuration for that specific device 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta128) Chapter 3 Bandwidth Management and Gall Admission Control, RSVP-Enabled Locations this variation, RSVP can be enabled selectively between location pairs, and the use of RSVP makes the CAC mechanism topology-aware, based on currently available bandwidth The RSVP-spabled location enables for ub-and-spok, filly meshed, or partal-meshed enviroamens automatically adapting to network changes, link failures, backup, and loa Sharing paths. ‘To configure locations perform the following: ‘Stop 1. Configure the RSVP service parameters using System > Service Parameters > Cisco CallManager, Step 2. Configure the RSVP Agents inthe router. Step. Add the RSVP Agents o CUCM using Media Resources > Media Termination Point ‘Stop 4. Enable RSVP between the location pairs using System > Location, Step 5. Configure Media Resource Groups (MRO) ‘Step 6. Configure Media Resource Group Lists (MRGL). ‘Step 7. Assign MRGLs to devices. Default intertocation RSVP po! es include the following: No Reserva © Optional (Video Desired) 1 Mandatory 1B Mandatory (Video Desired) Automated Alternate Routing (AAR) [AAR provides fallback mechanism fr calls denied by CAC and reroute alls over the PSTN automatically, being completely transparent othe user. orks only for CAC and fr cal place ineroal dicctory mumbcs. AAR does work with SRST and is not activated by WAN fair, To configure AAR, do the following 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta129) Chapter 3 Bandwidth Management and Gall Admission Control, ‘Stop 1. Use System > Service Parameters > Ciseo CallManager, Step 2. Configure the partitions and CSSs. ‘Stop 3. Configure AAR groups using Call Routing > AAR Groups Step 4. Configure phones for AAR. AAR Group configuration is not required with globalized call routing using local groups because prefixes are not required fr global acd dialing, SIP Precondition Also referred to as Interluster RSVP, SIP precondition applies to SIP trnks using RSVP for calls going out ofthe cluster. SIP precondition provided CAC between clusters at each end of the SIP trunk, SIP precondition can be used between CUCM clusters, CUCM Expres, gateways, and CUBE, RSVP-cnabled locations provide CAC within the CUCM cluster. SIP precondi multiple call-routing domains, 2m extends the use of RSVP beyond the cluster to ‘The configuration for SIP precondition isthe same as required for RSVP, plus configuring the SIP trunk: forthe SIP precondition using the SIP profile. Configure the SIP profile using the EE option for RSVP ower SIP, If both ends support SIP precondition and the reservation fils, no fallback to local RSWP occurs. However, fallback to local QoS is supported i SIP precondition isnot sup- ported atthe remote end H.323 Gatekeeper CAC 11323 gatekeeper provide address resolution, call routing, and CAC, which forthe gatekeeper is accomplished using the bandwidth command. The CUCM configuration provides the backup configuration using route lists and route group to provide redundant paths The following example provides an example of the gatekeeper for two clusters; CluserA and Cluster: 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta130] Chapter 3 Bandwidth Management and Gall Admission Control, gatekeeper yone pref 39 512" sone prafie REMOTE 513° in is example, the bandwidth command provides the CAC mechanism, The default configuration for interzone bandwidth is 160k, meaning that a total of 160 k of trafic willbe limited between all zones not defined by a more specific command for that zone. For example, the interzone bandwidth default command does not apply to the HQ zone, which is configured for 320k. ‘The bandwidth session limits the individual call to 16 k, meaning that all calls will be limited to G.729. ‘This configuration cam provide PSTN backup for calls rejected by CAC. Backup and redundant path are configured using Route Lists and Route Groups, To configure H, Step 1. Step 2. Step 3. Step 4. Step 5. Step 6. Gatekespers, do the following: "Enable the gatekeeper in Cisco 10S. ‘Ad the gatekeeper in CUCM. ‘Add the gatckoeper-contolled trunk Configure Route Groups, Route List (Configure the gatekeeper for CAC. “Modify the Route List and Route Groups to provide a PSTN backup path and Route Patterns othe gackeeper for CAC. 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate detaIst] GONP Voice GIPT2 642-457 Quick Reference, Second Edition Chapter 4 Applications and Features for Multisite Deployments This chapter reviews the various mobility features such as CUCM Device Mobility and Unified Communications Manager Extension Mobi Device Mobility This section reviews the configuration of Device Mobilit to enable mobile users o roam betwoen sites with uhir endpoints and have location-dependent settings applied In these cases, users may be wireless phones or Cisco Unified. IP Communicator. Overview Ifa phone is moved betwoen sites, the location dependent settings become inaccurate, Common stings that become inaccurate include region, location, SRST reference, AAR group, CSS, MRG, and MRGL, CUCM needs o be aware of the device location to adapt these settings based on the device location. Device Mobility provides this solution by triggering these changes based on the IP subnet ofthe specific device. Two types of phone configuration parameters can be dynamically assigned by Device Mobility ronming-senstive settings such as, regions, locations, Local Route Groups, Date/Time group, SRST, and MRGL. Device Mobiliy-rclted settings consist of Device- Mobility CSS, AAR CSS, and AAR group, 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta132] CChapter 4 Applications and Features for Mulisite Deployment LLocaton-dependent parameters are configured using device pools based onthe IP address of the phone. CUCM configures the phone with comect site-specific settings. Device Mobility Configuration Elements ‘The clements used are as follows Device Pool: Defines a common set of characteristics for devices 1 Device Mobility Info: Specifies an IP subnet and associates it with one or more device pools '© Physical Location: A tag assigned to one or more device pools; usd to identify whether the roam is betwcen physical loca- assigned to one or more device pools; used o identify whether the roam is between Device Operation of Device Mobility ‘When a Device Mobily enabled phone registers with CUCM wi Mobility Inf, the following occurs: ‘an address matching an IP subnet that is configured with Device 1 Ifthe Device Mobility Info is associated with the home device pool of the phone, the pho is considered to bein its home location; no reconfiguration occurs, 1 Ifthe Device Mobility Info is associated with one or more device pools other than the home device pool of the phone, one of the associated device pools is chosen, the physical locations are the same, the configuration of the phone is not modified. the physical locations are different, the roaming-sensitive parameters of the current device pool are applied, the Device Mobility Groups ar the same, the Device Mobliy-rclated settings ae also applied Im all other cases, the home device pool settings are applied. 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta133) CChapter 4 Applications and Features for Mulisite Deployment Device Mobility Configuration Follow thee steps to conigue Device Mobil Stop 1. ‘Stop 2. Configure Device Mobility Groups using System > Device Mobility > Device Mo ty Group, Step 3. Configure device pools Stop 4. Configure the Device Mobily Ino using System > Device Mobility > Device Mobility Info. ‘Step 5. Set the system default forthe Device Mobility mode using System > Service Parameter, choose the Cisco CallManager service and st the Device Mobility Mode to On or Off as required Step 6. Set the Device Mobility Mode for individual hones using the phone configuration window. Line and Device CSS ‘The Line CSS is never modified by Device Mobily. The Device CSS is modified only when the device roams beeen diferent plysical locations within the same Device Mobility Group. This provides the proper class of service and cll routing when using the Line/Device approach to panitions and calling search spaces. Device Mobility and Globalized Call Routing Without globalized routing, device mobility leads to improper routing of using the home gateway. In these cases, Device Mobility resuls in suboptimal routing when using diferent Device Mobility Groups. Unless TEHO is used, the caller must citer adapt othe local dialing rues of the remote location or use their home gateway, which results in suboptimal cll routing ‘The we of Local Route Groups. and globalized dal plan climinate thence for Device Mobily Groups enabling callers to se thet home dialing rales. The results optimized routing by updating the roaming-sensiivesetings 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta134) CChapter 4 Applications and Features for Mulisite Deployment Extension Mobility Extension Mobility enables CUCM users to lg in to an IP Phone and get thir personal profile applied regardless ofthe physical location. Issues Issues with using a guest phone at a remote site include the following 1 Extensions are bound to constant devices. ‘Th usr typically gels the wrong extension on that phone, ‘The usr gets the wrong calling privileges. ‘The user does not have his speed dal, ‘he usr has the wrong services ‘Thhe Message Waiting Indicator does not work properly Device profiles inthe Extension Mobility feature solve these issues by applying user-specific stings toa logged in phone, rather than deviee-specifc settings. User-specifc stings are applied in device profile, These device profiles are used to reconfigure the phone hen the user logs in to the phone enabled For extension mobility. Device-specific settings remain unchanged to provide the proper call routing, Overview ‘This feature works asa phone service in a CUCM cluster. If the user is sill logged in somewhere else, you cam allow muliple logins, deny the login, or perform auto-logout after a configurable maximum login time. Two types of configuration parameters are use 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta1H Userspecific, devieeevel parameter 138) CChapter 4 Applications and Features for Mulisite Deployment '© The configuration of phone buttons, including lines, speed dials, call park button and other options configured within the device profil Operation Here isthe sequence of evens that occurs when a user wants to log in oa phone using Extension Mobility Step 1. Step 2. Step 3. Step 4. “The user presses the Services button and chooses Extension Mobility. “The wser enters the User ID and PIN, “The proper device profil is chosen forthe ust. ‘The IP Phone resets and loads the proper configuration For phone model differences, the following occurs: Step 1. Step 2. Step 3. Step 4. Step 5. DDevice-dependent parameters from the default device profile are applied tothe phone, ‘The system copies all doviceindependent configuration settings from the device profile othe login device ‘The applicable device-dependent parameters of the user's device proile are applied Phone service subscriptions from the user's device profile are applied to the phone, Ifthe user's device profile docs not have phone services configured, the system uses the phone scrvies that are confige tured inthe default device profile ofthe login device. ‘The default device profile is applied only ifthe device mode! of the user's profile and the physical phone device being lo ged into are diferent model srice types for example, 7965 and 7971). When the phone mode! scrics isthe same (or example, 797x) between the plysical device and user profile, the Feature Safe Feature enabl thes differences. I feature safe is not supported, you must create ‘uliple default device profiles foreach model ina series or create multiple user profiles foreach phone model inthe serie, Remote tsces might want to use multiple profiles when traveling to different locations or as wanted, depending on their specific needs 122011 cisco Syeteme Ie Alright reared. The publction protected by copyright Place sae page 0 formate deta138) CChapter 4 Applications and Features for Mulisite Deployment For call routing the use of local groups is recommended when the traditional CSS approach to partitions and search spaces is used 10 ensure proper routing, Also, use the lineldevice approach to provide scalability and o simplify the dial plan for lage enterprise envi ronments, The linedevice approach, slong with local route groups, greatly simplilics the dial plan by eliminating the nced fr redun- dant configurations per location. Extension Mobility doesnot modify the device CSS or the automated alternate routing CSS. It does replace the line CSS or CSSs configured atthe phone with the ine CSS or CSSs configured atthe device profile ofthe logged-in use. Configuration To configore Extension Mobility, do the following: Step 4. Acta the Cio Extension Mobily Feature Service using Tools> Service Activation, Step 2. Set the Cisco Extension Mobility Service Parameters using System > Service Parameters Step 2. Add the Cisco Extension Mobility Phone Service using Device > Device Settings > Phome Services Step 4. Create Default Device Profiles sing Device > Device Settings > Defant Device Profle. This configuration en the administrator to rete the various poties for each phone model, Because oh Feature safe eat, this i note pied when using te same model pe for th physi pone and wer profes Stop 5. Create Devic Profiles using Device > Device Settings > Defult Device Profile This shuld he crete forthe cr Stop 6. Subscibe the Device Profile o Extension Mobility Phone Service wing the Subscribe/insubscibe Services fom Relate Links. The Enters Sabscrion feature in the Services enables fer ths sevice tobe used foal phones in the cluster ths allowin the adminstator or uso sip his tp. ‘Step 7. Assoc fe Users with Device Profiles using User Management > End User, ‘Step 8. Configure Phones for Cisco Extension Mobility in the Phone Configuration window, check the Enable Extension [Mobility check box to enable Cisco Extension Mobility ‘Step 9. Subscribe the Phone to Extension Mobility Phone Service; in the Phone Configuration window, use the related link ‘Subscribe/Uinsubscribe Services. You can use the Bulk Administration Tool (BAT) tool apply this configuration, Also, if the service is configured as such, the Extension Mobility can be configured as an Enterprise Subscription, which automatically subscribes all phones inthe cluster to this service (22011 Cisco Systeme ne Alrightsrecered The pubcetion Ie protected by copyright Pare ze page A formate deta137] GONP Voice GIPT2 642-457 Quick Reference, Second Edition Chapter 5 Call Control Discovery This chapter describes Service Advertisement Framework (SAF) and Call Control Discovery (CCD), which enable call agents to propagate and lear routes from the SAF-enabled network and thercby simplifying the dial plan for large enterprise deploy mens SAE enables services tobe propagated through a SAF-enabled network, whereas CCD isthe fist supported application for SAF, providing reachability for internal directory numbers and PSTN backup numbers. CCD then uses SAF to dynamically exchange cal routing information. Without SF, full-mesh networks do not scale and become complex. Hub-and-spoke networks provide beter scalability but quire ‘manual configuration and PSTN backup. In both cases, there is no dynamic exchange of callrouting information and no automatic SIN beckup. Gatekeeper provide the cll routing mechanism in these cases, SAE provides these features to large deploy mes, CCD, introduced in version 8x software, provides the client feature is available within the following: © Cisco Unified Communication Manager (CUCM) © Cisco Unified Communication Manager Express Cisco Unified SRST 18 Cisco Unified Border Element (CUBE) Cisco 10S gateways, Each CCD-enable cll agent advertises its local directory numbers and ranges, and any atached PSTN numbers or prefixes, SAF is ity to dynamically learn and advertise routs using SAF client. The SAP 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta138) ‘Ghapter 5 Call Control Discovery then used to propagate this information throughout the SAF-enabled network through SAF forwarders, ‘SAE forwarders lean information from SAF clits and advertise this information to other SAF forwarders in the SAF-enablednet= ‘work. In this way, SAF clients are aware of all learned call-routing information, Service Advertisement Framework (SAF) SAP isa network-based, salable, bandwidth-efficent, ealtime aprosch ta service advertisement and discovery SSAF focwanders are responsible to propagate services in the SAF-cnabled network. They do mt interpre the servic ts. uranes the fat reliable exchange of information, These SAF forwarders use the SAF Forwarding Protocol ($F FP) t forward information betncen SAF forwarders. SAF Forwarder, intra interact with SAF clients, These SAF clients are mits that actually process the sevice, which is adrised and lard rom te SAF forvarder. SAF cin ws the SAF Client Protcol (SAF CP) to commnicatto SAF forwarders. SAF cin ate various devices at previously described. However, SAF forwarders are always IOS gateways Following are two types of SAP clients '© Internal SAF clients: Devices in which the SAF client and forwarder are different devices. In this cae, the SAF client is cucM © Extemal SAF clients: Deviees in which the SAF client and forwarder functionality are colocated within the same device, such as CUCM, CUCM Expres, Cisco Unified SRST, and CUBE. ‘The SAF message consists ofthe SAF header (used by the SAF forwarder), and SAF service data (used by the SAF lien), The SAF header is similar tthe IP header ofa data frame, where the SAF service data canbe compared tothe TCP or UDP data frame ‘The SAF FP protocol is independent of the other routing protocols within the network, yet uses the same features and functions of the EIGRP protocol Like EIGRP, SAF FP uses neighbor relationships to advenise and ler routing information. However, the SAF protocol provides two options for this neighbor relationship: 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate deta139) ‘Ghapter 5 Call Control Discovery Layer adjacent Unicast or multicast '© Nor-layer 2 adjacent: Static configuration, which i used when one or more hops exist between forwarders, SAF clients perform the fllowing functions: 1B Register wo the network 1 Publish services Subscribe to services. 1 Send keepalives ‘SAF forwarders perform the following functions 1 Propagate updates from SAF clients to other SAF forwarders. 1 Send hellos to other SAF forwarders 1 Propagate updates to SAF cliens, Call Control Discovery (CCD) Overview SAP clin era along ihn ad wed ita erie das SA fred CD fe Cc en cal apna nna ea deo and SPN motor OCD ae cal ge Tor go SAP todutint hr inhematon ogi! te SAFeabledncvr. The SAF cas gem he SAF ence aa oroaang ‘inmost SA hoes Wie SAP cnbled evo Si cows echenge ar acis a Because SAF clon send only thei local information (directory numbers and PSTN numbers), CCD provides a more simplified di plan for the enterprise network. 122011 Cisco Systeme no lights recered The pubcetion Ie protected by copyright Pare zoe page A formate detaCCNP Voice CIPT2 642-457 Quick Reference, Second Edition Dave Schulz Technical Edtor: Alex Hannah Cope 20 ee i ay ‘hgh ae opt oth yl i Roce ‘atoepye rag ory my frontage anal =p, Seti pmo oe tie np fore se Wa Flow ay 201 ‘Warning and Disclaimer hs ditt uid ees pie inmate C2 eetC13 nay olen hema athe dst a Rlscce ‘Cleese pe bo worn on pi Suave ar ays opal ypc acl Si Go hte "a pn eee in itl Qui Rn bg te ath nd sect unt 140} GONP Voice GIPT2 642-457 Quick Reference, Second Edition “Trademark Acknowledgments Aas onlin i ita ued Rees tat ow be nds ai mith es Pec cpio Caco ro race Sea. cane sta ote a tan eran Ui cs Inistg Qit Reena nae madd afin ayo) nema ee ek Feedback Information ‘Cie Pes listo in-depth isi bok hig gai and ale Fk ok ld ith ‘rsslogo sis deropra at are nieexpe of wanef e peel ‘Shoal men (puyol uk eens or coe ser ta oer nde cs cto gh al ‘hack cnopen co Pane bre nie ial Gok Helene ie SBN our mee We yay ppc eur msi. Corporate and Government Sales “Th paella i pil Qk Reload guy or aku, {ronal ih ny nc lesen wir csr ce sad nate ot ey "ang sk meg ayant Fr mae ine, pe oar US. Cope and ‘iperinn Solr 8058251 cooper com cnet Ut Sit plan com Itratinal Salerno peso om (22011 Cico Sytem Ine lights reserved. This publication protcte by copyright