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The Cisco 2900 Series provides significantly enhanced modular capabilities offering investment protection for customers. Cisco 2900 series offer unparalleled total cost of ownership savings and network agility through the intelligent integration of market leading security, unified communications, wireless, and application services.The Cisco 2900 series ISRs offer embedded hardware encryption acceleration, voice- and video-capable digital signal processor (DSP) slots, optional firewall, intrusion prevention, call processing, voicemail, and application services.
The IP Communications High Density Voice Network Modules combine WAN Interface Card (WIC) and Voice Interface Card (VIC) functionality to provide unparalleled flexibility and power. The network modules provide enterprises, managed service providers, and service providers with the ability to directly connect devices to Cisco® 2600XM, Cisco 2691, 2811, 2821, 2851, Cisco 3700 Series and Cisco 3800 Series Access Routers for either IP Communications capabilities or pure toll bypass. The network modules can be integrated with these IP Telephony solutions in a Multiservice Access Router in order to provide a complete IP Communications solution for full service branch customers.
Cisco recommends that you have knowledge of basic Voice over IP (VoIP) concepts and configuration. Uses VWICs to supply physical interface (1 and 2 Port T1 Multi-Flex Trunk VWIC) (1 and 2 Port E1 Multi-Flex Trunk VWIC).
DSP IDs on the NM-HDV PVDM-12a€”When you configure the ds0-group or pri-group, the timeslots are assigned DSP channels dynamically.
Earlier versions of this chapter based the deployment models discussion exclusively on the call processing deployment models for Cisco Unified Communications Manager (Unified CM).
Table 10-1 lists the topics that are new in this chapter or that have changed significantly from previous releases of this document. From its beginnings with Voice over IP (VoIP) and IP Telephony 10 to 15 years ago, and continuing today with Unified Communications and Collaboration, users expect to be able to meet and communicate in a variety ways using a range of devices with differing capabilities. In general terms, the deployment model architecture follows that of the enterprise it is deployed to serve. In some cases, the deployment model of a technology will depart from that of the enterprise, due to technological constraints.
Another option for customers who exceed the sizing limits of a standard cluster is to consider deploying a megacluster, which can provide increased scalability. Note Unless otherwise specified, all information contained within this SRND that relates to call processing deployments (including capacity, high availability, and general design considerations) applies only to a standard cluster with up to eight call processing subscriber nodes. Where the Unified Communications and Collaboration services, their associated endpoints, gateways, border controllers, media resources, and other components are all located on a single high speed LAN or MAN. Where the Unified Communications and Collaboration services are located in a central campus site or data center, but the endpoints, gateways, media resources, and other components are distributed across multiple remote sites that are interconnected by a QoS-enabled WAN. There are an infinite number of variations on these three basic deployment models, such as deployments with centralized or distributed PSTN access and services, but the basic design guidance provided in this chapter still applies to the majority of them. Unified Communications services offer many capabilities aimed at achieving high availability. For services that are considered essential, redundant elements should be deployed so that no single point of failure is present in the design. In some instances, it is advantageous to deploy redundant IP links, such as IP WAN links, to guard against the failure of a single WAN link. Some products support the distribution of redundant service nodes across WAN links so that, if an entire site is off-line (such as would be the case during an extended power outage exceeding the capabilities of provisioned UPS and generator backup systems), another site in a different location can ensure business continuance. The capacities of various deployment models are typically integrally linked to the capacities of the products upon which they are based.
In this context, size generally refers to the number of users, which translates into a quantity of IP telephones, voice mail boxes, presence watchers, and so forth. These components are often considered adequate in the Local Area Network (LAN): QoS is achievable with all LAN equipment, bandwidth is typically in the Gigabit range, latency is minimal (in the order of a few milliseconds), and excellent reliability is the norm.
The Metropolitan Area Network (MAN) often approaches the LAN in all three dimensions: bandwidth is still typically in the multiple Megabit range, latency is typically in the low tens of milliseconds, and excellent reliability is common. The Wide Area Network (WAN) generally requires extra attention to these components: the bandwidth is at a cost premium, the latencies may depend not only on effective serialization speeds but also on actual transmission delays related to physical distance, and the reliability can be impacted by a multitude of factors. Bandwidth has great influence on the types of Unified Communications services available at a site, and on the way these services are provided. The influence of latency on design varies, based on the type of Unified Communications service considered for remote deployment. Reliability of the site's connectivity into the rest of the network is a fundamental consideration in determining the appropriate deployment model for any technology. Power loss is a very disruptive event in any system, not only because it prevents the consumption of services while the power is out, but also because of the ripple effects caused by power restoration.
Some Unified Communications services are delivered though the use of equipment such as servers that require periodical maintenance.
Throughout this document, design guidance is organized along the lines of the various Unified Communications services and technologies. Generally speaking, most aspects of any given Unified Communications service or technology are applicable to all deployments, no matter the site's size or network connectivity. For applications where enterprise branch sites are geographically dispersed and interconnected over a Wide Area Network, the Cisco Unified Communications services can be deployed at a central location while serving endpoints over the WAN connections. Centrally provisioned Unified Communications services can be impacted by WAN connectivity interruptions; for each service, the available local survivability options should be planned. The centralization of services need not be uniform across all Unified Communications services. In many cases, the main criteria driving the design for each service are the availability and quality of the IP network between sites. Also, when a given service is deployed centrally to serve endpoints at multiple sites, there are often advantages of feature transparency afforded by the use of the same processing resources for users at multiple sites. These advantages of feature transparency and economies of scale should be evaluated against the relative cost of establishing and operating a WAN network configured to accommodate the demands of Unified Communications traffic.
Unified Communications services can also be deployed independently over multiple sites, in a distributed fashion.
The main advantage of distributing Unified Communications services lies in the independence of the deployment approach from the relative availability and cost of WAN connectivity. If two sites are provisioned with independent services, they can still be interconnected to achieve some degree of inter-site feature transparency. To achieve such geographical diversity, the individual service must support redundant nodes as well as the deployment of these nodes across the latency and bandwidth constraints of the IP WAN.
Table 10-2 summarizes the ability of each Cisco Unified Communications service to be deployed in the manners outlined above.
Because call processing is a fundamental service, the basic call processing deployment models are introduced in this chapter. This section describes the fundamental deployment models for Cisco Collaboration and Unified Communications systems, and it lists best practices for each model. In this call processing deployment model, the Unified Communications services and the endpoints are co-located in the campus, and the QoS-enabled network between the service nodes, the endpoints, and applications is considered highly available, offering bandwidth in the gigabit range with less than 15 ms of latency end-to-end.
Alternatively for smaller deployments, Cisco Business Edition 6000 may be deployed in the campus. Maximum of 40,000 configured and registered Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP) IP phones, softphones, analog ports, video endpoints, SIP-based TelePresence endpoints and room-based TelePresence conferencing systems, mobile clients, and Cisco Virtualization Experience Clients (VXC) per Unified CM cluster.
Co-located digital signal processor (DSP) resources for conferencing, transcoding, and media termination point (MTP). Other Unified Communications services, such as messaging (voicemail), presence, and mobility are typically co-located. Interfaces to legacy voice services such as PBXs and voicemail systems are connected within the campus, with no operational costs associated with bandwidth or connectivity.
SIP-based video ISDN gateways are needed to communicate with videoconferencing devices on the public ISDN network.
Cisco Expressway-C and Cisco Expressway-E provide a collaboration edge function that enables secure business-to-business telepresence and video communications, and enterprise access for remote and mobile workers over the internet.
Cisco TelePresence Video Communication Server (VCS) may also be used to register legacy H.323 and third-party telepresence endpoints. High-bandwidth video (for example, 1.5 Mbps with 4CIF or 720p, to 2 Mbps with 1080p) is available between devices within the site. Ensure that the infrastructure is highly available, enabled for QoS, and configured to offer resiliency, fast convergence, and inline power. Implement the recommended network infrastructure for high availability, connectivity options for phones (in-line power), Quality of Service (QoS) mechanisms, and security.
In this call processing deployment model, endpoints are remotely located from the call processing service, across a QoS-enabled Wide Area Network. The IP WAN also carries call control signaling between the central site and the remote sites. Note In each solution for the centralized call processing model presented in this document, the various sites connect to an IP WAN with QoS enabled.
Cisco Business Edition 6000 may be deployed in centralized call processing configurations for up to 49 remote sites. Digital signal processor (DSP) resources for conferencing, transcoding, and media termination point (MTP) are distributed locally to each site to reduce WAN bandwidth consumption on calls requiring DSPs. Cisco Expressway-C and Cisco Expressway-E provide a collaboration edge function that enables secure business-to-business telepresence and video communications, and VPN-less enterprise access for remote and mobile workers over the internet. The system allows for the automated selection of high-bandwidth video (for example, 1.5 Mbps with 4CIF or 720p, to 2 Mbps with 1080p) between devices in the same site, and low-bandwidth video (for example, 384 kbps with 448p or CIF) between devices at different sites. A minimum of 1.5 Mbps or greater WAN link speed should be used when video is to be placed on the WAN.
For voice and video calls, automated alternate routing (AAR) provides the automated rerouting of calls through the PSTN when call admission control denies a call between endpoints within a cluster due to lack of bandwidth.
Call Forward Unregistered (CFUR) functionality provides the automated rerouting of calls through the PSTN when an endpoint is considered unregistered due to a remote WAN link failure. Cisco Unified Communications Manager Express (Unified CME) may be used for remote site survivability (Enhanced SRST) instead of an SRST router.
Cisco Unified Communications Manager Express (Unified CME) can be integrated with the Cisco Unity Connection server in the branch office or remote site.
With multisite centralized call processing model, PSTN routing through both central and remote site gateways is supported. Routers that reside at the WAN edges require quality of service (QoS) mechanisms, such as priority queuing and traffic shaping, to protect the voice and video traffic from the data traffic across the WAN, where bandwidth is typically scarce. Minimize delay between Unified CM and remote locations to reduce voice cut-through delays (also known as clipping). Configure Enhanced Locations CAC in Unified CM to provide call admission control into and out of remote branches.
The number of IP phones and line appearances supported in Survivable Remote Site Telephony (SRST) mode at each remote site depends on the branch router platform, the amount of memory installed, and the Cisco IOS release. If a distributed call processing model is deemed more suitable for the customer's business needs, the choices include installing a Unified CM cluster at each site or running Unified CME at the remote sites. SRST, Enhanced SRST, SIP SRST, SRSV, and MGCP Gateway Fallback can reside with each other on the same Cisco IOS gateway. When deploying Cisco Unified Communications across a WAN with the centralized call processing model, you should take additional steps to ensure that data and voice services at the remote sites are highly available. Getprice compares prices across all of your favourite products Australia-wide, covering all major cities including Sydney, Melbourne, Perth, Canberra and Brisbane. As the enterprise extends its IP telephony deployments from central sites to remote branch offices and teleworkers, a critical factor in achieving a successful deployment is the capability to support backup call control at these remote locations. Cisco Unified SRST or Unified E-SRST is a critical component of a centralized call-processing architecture in which a Cisco Unified Communications Manager cluster, located at a central site, provides telephony services for all sites of an organization.
A centralized call-processing architecture must include a strategy for survivability of telephony service at the remote locations (that is, at branch offices and the homes of teleworkers) when access to the centralized call-processing services is interrupted because of a WAN outage or other factors.
Cisco Unified Communications Manager or Cisco Business Edition in combination with Cisco Unified SRST or Unified E-SRST, which is embedded in the Cisco IOS® Software, helps provide high-availability IP telephony to remote locations. The enhanced reliability makes Cisco Unified Communications a cost-effective solution to help ensure telephony operation for all users in an organization, whether they are located in the headquarters or in a remote location.
Furthermore, in certain environments, the security of telephony communication is a critical requirement.
Cisco Powered Cloud Collaboration Service deployments powered by Cisco HCS are also a type of centralized call-processing architecture.
Cisco developed Cisco Unified SRST technology for all Cisco IOS Software platforms that support call processing (refer to Table 2 for a complete list of supported platforms).
Cisco Unified SRST functions in the remote-location router to automatically detect a failure in the network and initiate a process to provide call-processing backup redundancy for the IP phones in that location and help ensure that the telephony capabilities stay operational. Cisco routers running Cisco Unified SRST also offer a secure voice mode with Cisco Unified SRST.
Cisco Unified SRST supports Secure Client Control Protocol (SCCP) and Session Initiation Protocol (SIP) for Cisco IP Phones, providing basic telephony functions when the network SIP proxy or Cisco Unified Communications Manager is no longer available.
Based on Cisco Unified SRST and Cisco Unified SRST Manager, Cisco Unified E-SRST delivers all the benefits of Cisco Unified SRST along with improved management capabilities and better end-user experiences in survivability mode.
Cisco Unified SRST Manager is the management tool and is included as part of the Cisco Unified E-SRST solution at no extra charge. Cisco Unified SRST Manager resides at the central site and collects information from Cisco Unified Communications Manager or Cisco Business Edition. Cisco Unified SRST Manager operates within a virtual machine, running in the VMware ESXi (5.0 or later) hypervisor environment. For Cisco Unified E-SRST deployments, the branch office is configured in Cisco Unified Communications Manager Express-as-SRST mode on a Cisco Integrated Services Router (ISR). Figure 2 shows the supported topology model for Cisco Unified E-SRST at a branch-office site. Cisco Unified SRST and Unified E-SRST support 5 to 2000 phones on Cisco 800, 2900, 3900, and 4000 Series Integrated Services Router platforms.
Cisco offers ISR bundles with Cisco Unified SRST or Unified E-SRST at a discount when compared to separate purchase of bundle components. Cisco Unified SRST and Unified E-SRST are not dependent on Cisco Unified Communications Manager versions but on Cisco IP Phone loads. Table 5 lists the Cisco IP Phones using SIP that are supported by Cisco Unified SRST and Unified E-SRST. Table 6 summarizes the correlation between the Cisco Unified SRST version and Cisco IOS Software.


Secure Cisco Unified SRST for SIP phones is available with Cisco Unified SRST 8.0 and later versions.
The single-site model for IP telephony consists of a call processing agent located at a single site and a LAN or metropolitan area network (MAN) to carry voice traffic throughout the site. The multisite WAN model with centralized call processing consists of a single call processing agent that provides services for many sites and uses the IP WAN to transport voice traffic between the sites. Use this model for a main site with many smaller remote sites that are connected via a QoS-enabled WAN but that do not require full features and functionality during a WAN outage.
The multisite WAN model with distributed call processing consists of multiple independent sites, each with its own call processing agent connected to an IP WAN that carries voice traffic between the distributed sites. Use this model for a large central site with more than 30,000 lines or for a deployment with more than six large sites (more than 30,000 lines total) interconnected via a QoS-enabled WAN.
Use this model for a deployment with a maximum of six large sites (maximum of 30,000 lines total) interconnected via a QoS-enabled WAN.
NoteOther sections of this document assume that you understand the concepts involved with these deployment models, so please become thoroughly familiar with them before proceeding. The single-site model for IP telephony consists of a call processing agent located at a single site, or campus, with no telephony services provided over an IP WAN. Figure 2-1 illustrates the model for an IP telephony network within a single campus or site.
A single infrastructure for a converged network solution provides significant cost benefits and enables IP telephony to take advantage of the many IP-based applications in the enterprise.
The multisite WAN model with centralized call processing consists of a single call processing agent that provides services for many sites and uses the IP WAN to transport IP telephony traffic between the sites. NoteIn each solution for the centralized call processing model presented in this document, the various sites connect to an IP WAN with QoS enabled. Routers that reside at the WAN edges require quality of service (QoS) mechanisms, such as priority queuing and traffic shaping, to protect the voice traffic from the data traffic across the WAN, where bandwidth is typically scarce.
When deploying IP Telephony across a WAN with the centralized call processing model, you should take additional steps to ensure that data and voice services at the remote sites are highly available.
Under normal operations shown in the left part of Figure 2-3, the branch office connects to the central site via an IP WAN, which carries data traffic, voice traffic, and call signaling. The last two solutions in Table 2-1 use an ISDN backup link to provide survivability during WAN failures. With this option, ISDN is used for data survivability only, while SRST is used for voice survivability.
The centralized call processing deployment model can be adapted so that inter-site voice media is sent over the PSTN instead of the WAN. VoPSTN can be an attractive option in deployments where IP WAN bandwidth is either scarce or expensive with respect to PSTN charges, or where IP WAN bandwidth upgrades are planned for a later date but the IP telephony system is already being deployed.
The gateway port utilization resulting from these call forwarding flows should be taken into account when sizing the trunks connecting the branch to the PSTN.
To use the PSTN as the primary (and only) voice path, you can configure the call admission control bandwidth of each location (branch site) to be 1 kbps, thus preventing all calls from traversing the WAN. NoteIn this case, direct dialing of the shared line's DN from another branch would trigger multiple AAR-based PSTN calls. Abbreviated inter-site dialing can still be provided via a set of translations at each branch site, one for each of the other branch sites. A multisite WAN deployment with distributed call processing has many of the same requirements as a single site or a multisite WAN deployment with centralized call processing. Gatekeeper or Session Initiation Protocol (SIP) proxy servers are among the key elements in the multisite WAN model with distributed call processing.
SIP devices provide resolution of E.164 numbers as well as SIP uniform resource identifiers (URIs) to enable endpoints to place calls to each other. You can also use a combination of the two deployment models to satisfy specific site requirements.
These features make this solution ideal as a disaster recovery plan for business continuance sites or as a single solution for up to eight small or medium sites.
For clustering over the WAN to be successful, you must carefully plan, design, and implement various characteristics of the WAN itself. Jitter is the varying delay that packets incur through the network due to processing, queue, buffer, congestion, or path variation delay. The network should be engineered to provide sufficient prioritized bandwidth for all ICCS traffic, especially the priority ICCS traffic.
Provision the correct amount of bandwidth between each server for the expected call volume, type of devices, and number of devices. The network infrastructure relies on QoS engineering to provide consistent and predictable end-to-end levels of service for traffic.
The High Density Voice Network Module can support up to 60 simultaneous mid-complexity voice compression codecs or algorithms. Use the Software Advisor Tool (registered customers only) for a complete list of Cisco IOS software versions in which a particular feature, module, interface card, or chassis is supported. The current version of this chapter offers design guidance for the entire Cisco Unified Communications and Collaboration System, which includes much more than just the call processing service.
Today, a Unified Communications and Collaboration system could start with the deployment of Jabber Clients for IM and Presence only and incrementally add voice, video, web conferencing, mobile voice applications, social media, video conferencing, and telepresence as required. The main focus of this chapter is to provide the reader with design guidance for on-premises Collaboration deployments, but it also includes a description of systems such as Cisco Hosted Collaboration Solution (HCS) and Cisco WebEx, which can be deployed as managed cloud-based Collaboration services.
Deployment models describe the reference architecture required to satisfy the Unified Communications needs of well-defined, typical topologies of enterprises.
For example, if an enterprise has a single campus whose scale exceeds that of a single service instance, such as a call processing service provided by Cisco Unified Communications Manager, then a single campus might require more than a single instance of a call processing cluster or a single messaging product. Size also can be considered in terms of processing capacity for sites where few (or no) users are present, such as data centers. Packet treatment policies are generally available from MAN providers, so that end-to-end QoS is achievable. For example, if a site serving 20 users is connected with 1.5 Mbps of bandwidth to the rest of the system, the site's voice, presence, instant messaging, email, and video services can readily be hosted at a remote datacenter site.
If a voice service is hosted across a WAN where the one-way latency is 200 ms, for example, users might experience issues such as delay-to-dialtone or increased media cut-through delays. Pragmatic design decisions are required when balancing the need for reliability and the cost of achieving it.
A site with highly available power (for example, a site whose power grid connection is stable, backed-up by uninterruptible power supplies (UPSs) and by generator power) can typically be chosen to host any Unified Communications service. Some Unified Communications functions such as the hosting of Unified Communications call agent servers are best deployed at sites staffed with qualified personnel. For instance, the call processing chapter contains not only the actual description of the call processing services, but also design guidance pertaining to deploying IP phones and Cisco Unified Communications servers based on a site's size, network connectivity, and high availability requirements. For example, the call processing service can be deployed in a centralized manner, requiring only IP connectivity with the remote sites to deliver telephony services. As an example, the call processing service as offered by Cisco Unified CM can be configured with local survivability functionality such as Survivable Remote Site Telephony (SRST) or Enhanced SRST.
For example, a system can be deployed where multiple sites rely on a centralized call processing service, but can also be provisioned with a de-centralized (distributed) voice messaging service such as Cisco Unity Express. For example, when two sites are served by the same centralized Cisco Unified Communications Manager cluster, the users can share line appearances between the two sites.
For example, two sites (or more) can be provisioned with independent call processing Cisco Unified CME nodes, with no reliance on the WAN for availability of service to their co-located endpoints. For example, if a company operates a site in a remote location where WAN connectivity is not available, is very expensive, or is not reliable, then provisioning an independent call processing node such as Cisco Unified Communications Manager Express within the remote site will avoid any call processing interruptions if the WAN goes down. For example, a distributed call processing service provisioned through Cisco Unified Communications Manager Express can be inter-networked through SIP or H.323 trunks to permit IP calls between the sites. Depending on the design and features in use, this can provide the possibility for continued service during site disruptions such as loss of power, network outages, or even compromises in the physical integrity of a site by catastrophic events such as a fire or earthquake.
For example, the call processing service of Unified CM does support the deployment of a single cluster's call processing nodes across an IP WAN as long as the total end-to-end round-trip time between the nodes does not exceed 80 ms and an appropriate quantity of QoS-enabled bandwidth is provisioned.
For a detailed technical discussion on Cisco Unified Communications Manager call processing, refer to the chapter on Call Processing.
Likewise, the quality and availability of power are very high, and services are hosted in an appropriate data center environment.
Some campus call processing deployments may require more than one Unified CM cluster, for instance, if scale calls for more endpoints than can be serviced by a single cluster or if a cluster needs to be dedicated to an application such as a call center. Use the campus model if most of the calls from your enterprise are within the same site or to PSTN users outside your enterprise.
This practice eliminates the consumption of digital signal processor (DSP) resources for transcoding, and those resources can be allocated to other functions such as conferencing and media termination points (MTPs). Due to the limited quantity of bandwidth available across the WAN, a call admission control mechanism is required to manage the number of calls admitted on any given WAN link, to keep the load within the limits of the available bandwidth. Figure 10-2 illustrates a typical centralized call processing deployment, with a Unified CM cluster as the call processing agent at the central site and a QoS-enabled IP WAN to connect all the sites. Some centralized call processing deployments may require more than one Unified CM cluster, for instance, if scale calls for more endpoints than can be serviced by a single cluster or if a cluster needs to be dedicated to an application such as a call center. These resources may all be located at the central site or may be distributed to the remote sites if local conferencing resources are required. Connections to legacy voice services such as PBXs and voicemail systems can be made within the central site, with no operational costs associated with bandwidth or connectivity.
AAR relies on a gateway being available to route the call from the calling phone toward the PSTN, and another gateway to accept the call from the PSTN at the remote site, to be connected to the called phone. CFUR relies on a gateway being available to route the call from the calling phone toward the PSTN, and another gateway to accept the call from the PSTN at the remote site, to be connected to the called phone.
Video endpoints located at remote sites become audio-only devices if the WAN connection fails. The Cisco Unity Connection server is registered to the Unified CM at the central site in normal mode and can fall back to Enhanced SRST mode when Unified CM is not reachable, or during a WAN outage, to provide the users at the branch offices with access to their voicemail with MWI.
Providing a local gateway at a remote site for local PSTN breakout might be a requirement for countries that provide emergency services for users located at remote sites. When the IP WAN is down, or if all the available bandwidth on the IP WAN has been consumed, calls from users at remote sites can be rerouted through the PSTN. See the chapter on Call Admission Control, for details on how to apply this mechanism to the various WAN topologies.
SRST on a Cisco IOS gateway supports up to 1,500 phones, while Unified CME running Enhanced SRST supports 450 phones. Este router ofrece velocidades de banda ancha y la administracion simplificada para las pequenas empresas y sucursales de empresas pequenas y los teletrabajadores.
Cisco® Unified Survivable Remote Site Telephony (Unified SRST) and Cisco Unified Enhanced Survivable Remote Site Telephony (Unified E-SRST) provide cost-effective solutions for supporting redundant call control in remote branch offices and the homes of teleworkers. The architecture provides numerous benefits for enterprises, including centralized and simplified management. Call-processing redundancy in the remote location is particularly critical during an emergency (which may be the actual cause of the WAN outage). When access to Cisco Unified Communications Manager from a remote location is lost, for example, as a result of a WAN link failure, Cisco Unified SRST or Unified E-SRST provides telephony backup services to help ensure that the remote location has continuous telephony service. This solution supports secure telephony communication between any two phones in the network, whether those phones are in the headquarters facility or at a remote location.
Based on Cisco Unified Communications Manager, Cisco HCS offers industry-leading collaboration technologies for secure and scalable clouds from partners certified to offer Cisco Powered Cloud Services.
Cisco Unified SRST integrates network intelligence into Cisco IOS Software, which acts as the call-processing engine for IP phones located in the remote locations during a WAN outage (Figure 1).
Upon restoration of WAN connectivity, the system intelligently and automatically shifts call processing back to the primary Cisco Unified Communications Manager cluster. If secure voice is deployed with Cisco Unified Communications Manager at the central site, secure Cisco Unified SRST allows you to keep calls secure during Cisco Unified SRST mode with Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) for signaling and media encryption.
The Cisco Unified SRST router uses the proprietary SCCP protocol to register the SCCP phones and the SIP registrar services to support SIP phones. It enables automatic provisioning of branch-office routers and provides a richer telephony experience during failover mode by auto-provisioning the branch-office routers with features such as hunt groups, call park and pickup, and an ephone template from the centralized Cisco Unified Communications Manager.
Cisco Unified SRST Manager collects configuration information required for advanced features such as hunt groups and pickup groups, and distributes the configuration information to the branch-office sites. The software is packaged as an open-virtualization-archive (OVA) template for installation within the virtual-machine environment.
Cisco Unified E-SRST provisions the branch-office site using the Cisco Unified SRST Manager with information such as phone users, MAC addresses, and advanced telephony features configured on Cisco Unified Communications Manager or Cisco Business Edition. Table 4 lists the Cisco IP Phones supported by Cisco Unified SRST and Unified E-SRST with SCCP phone loads.
The IP WAN in this model does not carry call control signaling between the sites because each site has its own call processing agent. An enterprise would typically deploy the single-site model over a LAN or metropolitan area network (MAN), which carries the voice traffic within the site.
Depending on conferencing requirements, these resources may be either SCCP or H.323, or both. 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. This practice eliminates the consumption of digital signal processor (DSP) resources for transcoding, and those resources can be allocated to other functions such as conferencing and Media Termination Points (MTPs). Depending on conferencing requirements, these resources may be either SCCP or H.323, or both, and may all be located at the central site or may be distributed to the remote sites if local conferencing resources are required. These gateways may all be located at the central site or may be distributed to the remote sites if local ISDN access is required. Video is not recommended on WAN connections that operate at speeds lower than 768 kbps. If the WAN connection fails, SCCP video endpoints located at the remote sites become audio-only devices, and H.323 video devices fail.
In addition, a call admission control scheme is needed to avoid oversubscribing the WAN links with voice traffic and deteriorating the quality of established calls.


When the IP WAN is down, or if all the available bandwidth on the IP WAN has been consumed, users at the remote sites can dial the PSTN access code and place their calls through the PSTN. See the chapter on Call Admission Control, page 9-1, for details on how to apply this mechanism to the various WAN topologies. Table 2-1 summarizes the different strategies for providing high availability at the remote sites.
These solutions apply to both data and voice services, and are entirely transparent to the call processing layer. The branch router queries the IP phones for their configuration and uses this information to build its own configuration automatically. The branch router deletes its information about the IP phones and reverts to its standard routing or gateway configuration.
RDNIS is required so that calls redirected to voicemail carry the redirecting DN, to ensure proper voicemail box selection.
This includes the quantity of digits delivered in the call directories such as Missed Calls and Received Calls.
If MoH servers are deployed at the central site, then only calls placed on hold by devices at the central site will receive the hold music. AAR provides transparent re-routing over the PSTN of inter-site calls when the locations mechanism for call admission control determines that there is not enough available WAN bandwidth to accept an additional call. With this configuration, all inter-site calls trigger the AAR functionality, which automatically re-routes the calls over the PSTN. When bandwidth becomes available to support voice media over the WAN, the dial plan can be maintained intact, and the only change needed is to update the location bandwidth value for each site. If the destination phone has registered with an SRST router, then it can be reached by directly dialing its PSTN DID number. To preserve abbreviated dialing functionality under these conditions, configure the SRST router with an appropriate set of translation rules.
Depending on conferencing requirements, these resources may be either SCCP or H.323, or both, and may all be located at the regional sites or may be distributed to the remote sites of each cluster if local conferencing resources are required. These gateways may all be located at the regional sites or may be distributed to the remote sites of each cluster if local ISDN access is required.
Automated alternate routing (AAR) is also supported for both intra-cluster and inter-cluster video calls. Typically, the PSTN serves as a backup connection between the sites in case the IP WAN connection fails or does not have any more available bandwidth. Follow the best practices from these other models in addition to the ones listed here for the distributed call processing model.
They each provide dial plan resolution, with the gatekeeper also providing call admission control.
The main factors, for the purpose of design, are the size of the site and the functionality required. The design of each site varies with the call processing agent, the functionality required, and the fault tolerance required. For example, two main sites may each have primary and backup subscribers, with another two sites containing only a primary server each and utilizing either shared backups or dedicated backups at the two main sites. Delay for other ICCS traffic should be kept reasonable to provide timely database and directory access.
Jitter for the IP Precedence 3 ICCS traffic must be minimized using Quality of Service (QoS) features. This bandwidth is in addition to any other bandwidth for other applications sharing the network, including voice and video traffic between the sites.
Neither QoS nor bandwidth alone is the solution; rather, QoS-enabled bandwidth must be engineered into the network infrastructure.
There is also a real-time protocol called Intra-Cluster Communication Signaling (ICCS), which provides the communications with the Cisco CallManager Service process that is at the heart of the call processing in each server or node within the cluster.
The SQL database is replicated from the publisher server to all other servers in the cluster using best-effort. ICCS uses a Transmission Control Protocol (TCP) connection between all servers that have the Cisco CallManager Service enabled.
A tightly integrated Unified Communications architecture is required as the number of devices and forms of communication available to a single Unified Communications user increases.
The choice of using an on-premises, cloud-based, or hybrid solution may be determined by many factors. For example, a centralized call processing deployment model caters to enterprises whose operational footprint is based on multiple sites linked to one or few centralized headquarters offices. For example, the clustering technology used in Cisco Unified Communications Manager (Unified CM) allows for up to three servers to provide backup for each other.
For some of the products supporting services covered in more detail in other sections of this document, the capacities of those products are discussed in their respective sections.
If that same site is hosting 1000 users, some of the services would best be hosted locally to avoid saturating the comparatively limited bandwidth with signaling and media flows.
If a site has inconsistent power availability, it would not be judicious to use it as a hosting site. Likewise, the call admission control chapter focuses on the technical explanation of that technology while also incorporating site-based design considerations. Likewise, voice messaging services, such as those provided by the Cisco Unity Connection platform, can also be provisioned centrally to deliver services to endpoints remotely connected across an IP WAN. Likewise, a centralized voice messaging service such as that of Cisco Unity Connection can be provisioned to allow remote sites operating under SRST to access local voicemail services using Unity Connection Survivable Remote Site Voicemail (SRSV). Likewise, a Unified Communications system could be deployed where call processing is provisioned locally at each site through Cisco Unified Communications Manager Express, with a centralized voice messaging service such as Cisco Unity Connection. This benefit would not be available if each site were served by different (distributed) call processing systems. Likewise, sites can be provisioned with independent voice messaging systems such as Cisco Unity Express.
Likewise, separate instances of Cisco Unity Connection or Cisco Unity Express can partake in the same messaging network to achieve the routing of messages and the exchange of subscriber and directory information within a unified messaging network.
By contrast, Unified CME does not offer redundancy, and thus cannot be deployed in a geographically diverse configuration.
Communications between the endpoints traverses a LAN or a MAN, and communications outside the enterprise goes over an external network such as the PSTN.
In this deployment model, other Unified Communications services such as voice messaging, presence and mobility are often hosted at the central site as well to reduce the overall costs of administration and maintenance. Connectivity to legacy systems located at remote sites may require the operational expenses associated with the provisioning of extra WAN bandwidth.
In this case, the local gateway at the remote site provides call routing to the local PSAP for emergency calls.
For centralized call processing deployments, Enhanced Location CAC or RSVP-enabled locations configured within Unified CM provide call admission control (CAC).
The Cisco Unified Survivable Remote Site Telephony (SRST) feature, available for both SCCP and SIP phones, provides call processing at the branch offices for Cisco Unified IP Phones if they lose their connection to the remote primary, secondary, or tertiary Unified CM or if the WAN connection is down.
Both Cisco Unified SRST and Unified E-SRST support Cisco Unified Communications Manager and Cisco Business Edition.
Cisco Unified SRST contributes to this secure telephony communication solution by supporting the same secure telephony protocols in a remote location when that location loses communication with the centralized Cisco Unified Communications Manager. Cisco HCS is located at the providera€™s data center with the enterprises it serves acting like remote-office locations. The Cisco Unified SRST configuration needs to be completed only once, during the initial installation, simplifying deployment, administration, and maintenance. When the WAN link or Cisco Unified Communications Manager service is restored, Cisco Unified Communications Manager resumes secure call-handling capabilities.
Cisco Unified SRST offers fault monitoring using Simple Network Management Protocol (SNMP) with the SRST MIB, which allows you to remotely monitor the Cisco Unified SRST site using existing SNMP tools or CiscoWorks. If a WAN outage occurs, when the Cisco Unified E-SRST service running on the branch-office routers takes over call processing, it applies the configuration provisioned by Cisco Unified SRST Manager to provide enhanced telephony services at the branch-office sites.
The OVA file includes the Cisco Unified SRST Manager software, as well as the virtual-machine system settings preconfigured to operate with Cisco Unified SRST Manager. If an IP WAN is incorporated into the single-site model, it is for data traffic only; no telephony services are provided over the WAN. In this model, calls beyond the LAN or MAN use the public switched telephone network (PSTN). There is no dependency for service in the event of an IP WAN failure or insufficient bandwidth, and there is no loss of call processing service or functionality. H.323 might be required to support specific functionality such as support for Signaling System 7 (SS7) or Non-Facility Associated Signaling (NFAS). An active SCCP video call between two devices at remote sites will continue with both video and audio channels, but it will not be able to activate additional features such as call transfer. The Survivable Remote Site Telephony (SRST) feature on Cisco IOS gateways provides call processing at the branch offices in the event of a WAN failure. The choice of one of these strategies may depend on several factors, such as specific business or application requirements, the priorities associated with highly available data and voice services, and cost considerations.
The options range from adding a redundant IP WAN link at the branch router to adding a second branch router platform with a redundant IP WAN link. The branch router or gateway forwards both types of traffic (call signaling and voice) transparently and has no knowledge of the IP phones. The branch IP phones can then make and receive calls either internally or through the PSTN. Note that the Cisco VT Camera Wideband Video Codec is not supported over intercluster trunks. A site connected only through the PSTN is a standalone site and is not covered by the distributed call processing model. The bandwidth provisioned must have QoS enabled to provide the prioritization and scheduling for the different classes of traffic. The SQL traffic may be re-prioritized in line with Cisco QoS recommendations to a higher priority data service (for example, IP Precedence 1 if required by the particular business needs).
Cisco's Unified Communications and Collaboration architecture has the flexibility and scale to meet the demands of a rapidly changing and expanding Unified Communications environment that will become more URI-centric as users with multiple Unified Communications devices wish to be identified by a single user name irrespective of the form of communication. For example, cloud-based solutions require less on-site expertise but might lack the deployment flexibility that many enterprises need.
Another alternative is to consider increasing the bandwidth to allow services to be delivered across the WAN from a remote datacenter site. An enterprise would typically deploy the campus model over a single building or over a group of buildings connected by a LAN or MAN. Communication outside the enterprise goes over an external network such as the PSTN, through a gateway or Cisco Unified Border Element (CUBE) session border controller (SBC) that can be co-located with the endpoint or at a different location (for example, when using a centralized gateway at the main site or when doing Tail End Hop Off (TEHO) across the enterprise network).
In situations where the availability of the WAN is unreliable or when WAN bandwidth costs are high, it is possible to consider decentralizing some Unified Communications services such as voice messaging (voicemail) so that the service's availability is not impacted by WAN outages. Local PSTN breakout at remote sites might also be required for countries having strict regulations that require the separation of the IP telephony network from the PSTN. Cisco Unified SRST is available on Cisco IOS gateways or on Cisco Unified CME running Enhanced SRST. Cisco Unified SRST also provides reliable cloud communications and supports Cisco Powered Cloud Collaboration Services powered by Cisco Hosted Collaboration Solution (HCS). If communications from the Cisco Powered Cloud Service to the enterprise fail, Cisco Unified SRST provides the telephony backup service to help eliminate business disruption as described throughout this document. The Cisco SRST MIB provides the network operations center details about Cisco Unified SRST activity, including duration of Cisco Unified SRST use, IP phones registered or registration failure, and calls processed during SRST mode. Propagation delay between two sites introduces 6 microseconds per kilometer without any other network delays being considered.
The retransmission might result in a call being delayed during setup, disconnect (teardown), or other supplementary services during the call. An example of this is extensive use of Extension Mobility, which relies on SQL database configuration. Because only eight servers may have the Cisco CallManager Service enabled in a cluster, there may be up to seven connections on each server.
Hybrid designs can also be deployed, where some Unified Communications functions such as call control are provided on-premises and others are provided as a cloud-based service. For example, a phone may have as its first three preferred call control agents, three separate Unified CM servers belonging to the same call processing cluster. Where regulations allow, local PSTN breakout through the remote site gateway can be used to enable toll bypass or tail-end hop off (TEHO). Unified CME running Enhanced SRST provides more features for the phones than SRST on a Cisco IOS gateway. A backup WAN link connection is required to receive Cisco SRST MIB data at the central site in SRST mode.
Applications such as voicemail and Interactive Voice Response (IVR) systems are typically centralized as well to reduce the overall costs of administration and maintenance. This traffic is priority ICCS traffic and is marked dependant on release and service parameter configuration. As a fourth choice, the phone can also be configured to rely on a Cisco IOS router for call processing services. The various types of ICCS traffic are described in Intra-Cluster Communications, which also provides further guidelines for provisioning.
These distances are provided only as relative guidelines and in reality will be shorter due to other delay incurred within the network.



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