Implementing Intercluster Lookup Service

Save this PDF as:
 WORD  PNG  TXT  JPG

Size: px
Start display at page:

Download "Implementing Intercluster Lookup Service"

Transcription

1 Appendix 11 Implementing Intercluster Lookup Service Overview When using the Session Initiation Protocol (SIP), it is possible to use the Uniform Resource Identifier (URI) format for addressing an end user. Intercluster URI calls are routed based on the host part of a URI. Therefore, each cluster must use a different URI host part. The Intercluster Lookup Service (ILS), a feature that was introduced in Cisco Unified Communications Manager version 9, provides a mechanism that allows the same host part to be used in URIs across clusters. This lesson explains how ILS works, describes its main components, and how to implement ILS. Upon completing this lesson, you will be able to describe the purpose of ILS, its functions, and how to implement it. This ability includes being able to meet these objectives: Describe the purpose of ILS and the services it provides Describe the components of ILS networking and their functions Describe how URI syncing works and how it interacts with URI routing Describe what needs to be considered when implementing ILS Implement and monitor ILS operation

2 ILS Overview This topic provides an overview of ILS, a feature that was introduced in Cisco Unified Communications Manager version 9. Other clusters are reached based on SIP route patterns that match the host portion of the called URI. URI routing requires unique host portions per cluster amer.cisco.com emea.cisco.com Cisco and/or its affiliates. All rights reserved. COLLAB90 v Before Cisco Unified Communications Manager version 9, URI routing was based on the host part (the domain) of a URI. A SIP route pattern was configured for each domain, and this route pattern referred to a SIP trunk. The SIP trunk then pointed to one or more servers of the remote cluster. In order to route URIs successfully, the host part of URIs must be unique to each cluster. The figure shows an example with three Cisco Unified Communications Manager clusters. Each of them uses a dedicated domain (amer.cisco.com, emea.cisco.com, and apac.cisco.com) for the host part of its local URIs. Each cluster has two SIP route patterns and two SIP trunks that refer to the other two sites. This is a typical implementation of intercluster URI routing Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

3 Flat Domain Scheme Causes Problems with Multicluster URI Routing In some multicluster deployments, customers want to use a single domain for all URIs across all clusters. If URI host portions are not unique across clusters, URI routing does not work: No cluster identification is possible. Calls to remote destinations will fail Cisco and/or its affiliates. All rights reserved. COLLAB90 v The type of configuration that is shown in the figure causes problems with URI routing because there is no information in the URI that enables a Cisco Unified Communications Manager to find out the location of the endpoint. If the cluster where the target device is registered is not known, Cisco Unified Communications Manager cannot route the call. Consequently, a flat domain scheme or any overlapping of domains across clusters are not supported for URI routing in earlier versions than Cisco Unified Communications Manager version Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-3

4 ILS Solves Multicluster URI Routing Issues with Overlapping Domains With Cisco Unified Communications Manager version 9, ILS can be used to solve URI routing issues when using overlapping domains in the host part of directory URIs. Each cluster is configured with a globally unique route string. Each cluster propagates its local URIs and its local route string to all other clusters. Each cluster learns all remote URIs and a route string that is associated with each URI. Call routing is not based on the URI host portion but on a learned route string. This allows the same URI host portion (domain) to be used at multiple clusters. SIP route patterns: amer.cisco.com emea.cisco.com route string: amer.cisco.com ILS URI exchange: -> amer.cisco.com -> emea.cisco.com route string: emea.cisco.com 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS provides a mechanism to propagate individual URIs. In addition to the URI, each cluster adds a tag that uniquely identifies the cluster. This tag is called the route string, and it is configured by the administrator. In most cases, a cluster-specific subdomain of the enterprisewide domain is used. Through the propagation mechanism, all clusters learn all URIs and the associated route string. When a call is placed to a URI and the full URI is found in the ILS routing table, then a SIP route pattern for the associated route string is searched instead of the domain part of the called URI. In summary, ILS decouples the call routing information from the URI that is actually called and thus allows the same URI host part to be used across clusters. ILS is a new Cisco Unified Communications Manager feature service. This service builds on the existing SIP route pattern constructs. Other than Service Advertisement Framework (SAF) and Call Control Discovery (CCD), ILS does not eliminate the need to configure SIP trunks. It only modifies the process of choosing a configured SIP route pattern and the associated SIP trunk. Note Although SAF and CCD provide similar services (dynamic discovery of remote cluster information), there are many differences. For example, SAF and CCD do not support routing of alpha URIs, and ILS does not eliminate the local configuration of call routing information. ILS requires SIP route patterns and SIP trunks to be statically configured. The figure shows a multicluster deployment with two clusters. URIs in both clusters use the same domain for the host part of the URIs. Each cluster has a unique route string 11-4 Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

5 (amer.cisco.com and emea.cisco.com). Each cluster has a SIP route pattern that is configured. This SIP route pattern points to a SIP trunk, which then refers to the other cluster. This part of the configuration is static and is stored in the Cisco Unified Communications Manager configuration database. In addition, ILS is enabled at the two clusters and therefore each cluster advertises its local URIs along with the locally configured route string. In the example, the cluster shown on the left advertises URI with route string amer.cisco.com. The cluster shown on the right advertises URI with route string emea.cisco.com. After synchronization, each cluster has a URI that is applied to a local phone and a URI that is learned via ILS. When a call is placed to a remote URI, Cisco Unified Communications Manager finds the URI in the ILS URI routing table and looks up the associated route string instead of the host part of the called URI. The route string matches the statically configured SIP route pattern and is then routed out via the associated SIP trunk Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-5

6 Main ILS Components This section lists and describes the main components of ILS. Component ILS Networking URI Syncing Description Joins clusters into a network. Provides discovery of ILS services available on remote clusters. An ILS service that propagates and learns URIs and associated route strings. Propagation: Each cluster advertises all locally configured URIs and the locally configured route string to all other clusters. Learning: Each cluster listens to the advertisements that are sent by other clusters and builds a table of learned URIs and their associated route strings Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS consists of two main components: ILS networking: ILS networking joins clusters into a network. ILS networking provides a discovery of ILS services that are available on remote clusters. ILS services can utilize the ILS network to propagate service-specific information across clusters. URI syncing: URI syncing is an ILS service that propagates and learns URIs and their associated route strings throughout the members of an ILS network. Each cluster advertises all locally configured URIs and the locally configured route string of the cluster to all other clusters. Each cluster listens to the advertisements that are sent by other clusters and builds a table of learned URIs and their associated route strings. URI syncing is a service that interacts with other clusters and therefore is listed at the Cluster View page in Cisco Unified Communications Manager Administration. Other remote services that are shown on the Cluster View page include Location Bandwidth Manager (LBM), User Data Service (UDS), and services that relate to Cisco Extension Mobility Cross Cluster (EMCC) such as public switched telephone network (PSTN) access, Resource Reservation Protocol (RSVP) agent, and TFTP. When comparing SAF and CCD to ILS, ILS networking provides the functionality of SAF, and URI syncing provides the functionality of CCD Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

7 ILS Networking This topic describes ILS networking topologies. Clusters can be joined together to form an ILS network. Each cluster in an ILS network has a hub or spoke role. Each hub syncs directly with all other hubs (full mesh between hubs) and with its own spokes. A hub can have no spokes, one spoke, or more than one spoke. Each spoke has one hub and syncs directly with only this hub. Hub1, One Spoke Hub3, No Spokes Spoke2 Spoke1 Hub2, Two Spokes Full Mesh of Three Hub Clusters Spoke Cisco and/or its affiliates. All rights reserved. COLLAB90 v Cisco Unified Communications Manager clusters can be joined together to form an ILS network. Each cluster in an ILS network has one of the following roles: hub or spoke. Each hub synchronizes information directly with all other hubs and with its own spokes. A hub can have no spoke, one spoke, or more than one spoke. Each spoke has one hub and synchronizes information with only this hub. In summary, hubs are fully meshed, and spokes form a star topology with their hub. Any combination is possible: a single hub with associated spokes, multiple hubs with no spokes, or a mixed topology, as shown in the figure. By default, each cluster is in standalone mode. This means that the cluster is not part of an ILS network Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-7

8 ILS Synchronization This section describes how ILS synchronization occurs within an ILS network. Synchronization in an ILS network is pull-based. Sync interval is configured at each cluster (default 10 minutes, minimum 1 minute, and maximum 24 hours). Per interval, cluster pulls in information from every other cluster that is directly associated. - Hubs pull information from all other hubs. - Spokes pull information from their hub. Convergence time depends on the synchronization path (number of hops) and configured sync intervals. - Maximum distance in an ILS network is three hops: spoke hub hub spoke. - Maximum convergence time is the total of sync intervals along the synchronization path (by default, 30 minutes). Spoke1 Hub1 Hub2 Spoke1 Pull Hub1 Pull Hub2 Pull Spoke Cisco and/or its affiliates. All rights reserved. COLLAB90 v Synchronization in an ILS network is pull-based. This means that each cluster requests information from its connected clusters. Spokes pull information only from their hub. Hubs pull information from all other hubs. The synchronization interval can be configured independently at each cluster. The default is 10 minutes, and the configurable range is from 1 minute to 1440 minutes (24 hours). Because each cluster can be configured with different synchronization intervals, update times can also be different. The maximum update time is the total of all synchronization intervals along the synchronization path. The maximum synchronization path is determined by the ILS network topology. Because all hubs are fully meshed, the maximum number of hops in every ILS network is three. The figure shows this topology. Spoke1 is connected to Hub1. Hub1 is connected to Hub2, which is connected to Spoke2. When calculating the maximum update time for updates that occur in Spoke2 to be seen at Spoke1, the synchronization intervals of Spoke1, Hub1, and Hub2 are relevant. The synchronization interval of Spoke 2 is not applicable in this case. Note Although the synchronization interval can be configured independently per cluster, it is recommended that you configure the same synchronization interval at all members of an ILS network Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

9 ILS Networking: Operation Within a Cluster This section describes ILS networking operation within a cluster. ILS is a feature service that is activated per server. The recommendation is to activate ILS on all servers of a cluster. One server per cluster is designated as the cluster xnode: - The xnode is automatically selected and cannot be configured. - The xnode manages most of the communication with other clusters. Hub1 Hub3 Spoke2 Spoke1 Spoke3 Hub Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS is a feature service that can be activated or deactivated on each server within a cluster. It is recommended that you activate ILS on all servers of a cluster that joins an ILS network. One server per cluster is designated as the cluster xnode. The selection of the xnode is performed automatically and cannot be influenced by the administrator. Out of the servers that are running the ILS service, the subscriber server with the lowest node ID is designated as the xnode of the cluster. If the ILS service is only activated on the publisher, then the publisher is designated as the xnode. In a Cisco Unified Communications Manager cluster deployment (that is, when subscribers are present), it is not recommended that you activate the ILS service at the publisher only. The xnode manages most of the communication to other clusters Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-9

10 URI Syncing This topic describes how URI syncing works and how URIs that are learned via ILS are used for URI routing. URI syncing propagates and learns URIs and associated route strings. It can be configured on clusters that are members of an ILS network. It uses ILS network synchronization capabilities. Each cluster builds a complete list of URIs and associated route strings. - The list is stored in memory, not in the configuration database. - The list is used by the URI routing process if no local match is found: Is user portion of URI a DN? No Does URI match an entry in any accessible partition of local URI table? No Does URI match any learned ILS URI? No Does host portion of URI match a SIP route pattern in any accessible partition? No Block call. Yes Yes Yes No Yes Route as a DN. Offer call. Does route string of matched ILS URI match a SIP route pattern in any accessible partition? Yes Route using matching SIP route pattern Cisco and/or its affiliates. All rights reserved. COLLAB90 v URI syncing learns and propagates URIs and the associated route strings. URI syncing can be enabled on clusters that are part of an ILS network. URI syncing utilizes ILS network synchronization capabilities to exchange its data across all participating clusters. URI syncing follows the same syncing topology and syncing intervals as ILS networking. For example, a spoke syncs URIs directly only with its hub and indirectly with the rest of the network via its hub. Although URI syncing is configured cluster by cluster, a deployment in which URI syncing is intended, but is not configured on every cluster in the ILS network, is not recommended because it may easily result in errors. For example, if URI syncing is enabled on a spoke but not on its hub, the spoke will not be able to sync URIs with any other clusters. URI syncing enables each cluster to build a complete list of URIs and associated route strings. These ILS URI routing entries are stored only in the memory of the ILS servers. They are not stored in the cluster configuration database. The figure indicates how the ILS URI routing table is used by the URI routing process: If the user part of a URI is a directory number, then the number is looked up in the numeric call routing table. Note This lookup considers all entries of the numeric call routing table that are accessible to the calling device based on the calling search space (CSS) of the calling device and the partitions that are assigned to the routing table entries. If the user part of a called URI is not a directory number, then the full URI is looked up in the URI routing table, which contains all locally configured URIs (URIs that are applied to phones). If a match is found, then the call is sent to the locally registered device Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

11 Note This lookup is also following the CSS and partition logic. Like directory numbers, locally configured URIs can be assigned a partition. There is no separate CSS for URI routing. The CSS that is used for numeric call routing is also applicable to URI routing. If no match is found, then the full URI is looked up in the ILS URI routing table. Note The lookup in the ILS URI routing table always considers all entries of the ILS URI routing table. ILS URI routing table entries do not have a partition that is assigned, and the CSS and partition logic do not apply to ILS URI routing table lookups. If a match is found in the ILS routing table, then the associated route string is matched against all accessible SIP route patterns. If no matching SIP route pattern is found, then the host part of the originally called URI is looked up against all SIP route patterns. If the full URI was not found in the ILS URI routing table, then the host part of the originally called URI is matched against all SIP route patterns. Note SIP route patterns can be assigned a partition. When a route string or the host part of a called URI is matched against the SIP route patterns, the CSS and partition logic applies: SIP route patterns in partitions that are not included in the CSS of the calling devices are ignored. If a matching SIP route pattern is found, then the call is routed using the matched SIP route pattern. If no matching SIP route pattern is found, then the call fails Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-11

12 URI Import and Export This section describes the option to import and export URIs into Cisco Unified Communications Manager. URIs can be imported from third-party systems, such as VCS and OCS: Each import is stored in a directory URI-imported catalog. The Cisco Unified Communications Manager administrator assigns a route string to each directory URI-imported catalog. URIs and associated route strings of directory URI imported catalogs are treated as if they were learned via ILS URI syncing. URI-imported catalogs are also propagated to other clusters on the next pull. Local URIs can be exported: They can be exported and then imported on third-party systems that support URI import. ILS-learned URIs and imported URIs cannot be exported. To export all URIs of all clusters, export local URIs at each cluster Cisco and/or its affiliates. All rights reserved. COLLAB90 v URIs can be manually imported from third-party systems such as Cisco TelePresence Video Communications Server (VCS) and Microsoft Office Communications Server (OCS) or Microsoft Lync server. Each import is stored in a directory URI-imported catalog. The Cisco Unified Communications Manager administrator assigns one route string to each directory URI-imported catalog. Imported URIs and their associated route strings are treated as if they were learned via ILS URI syncing. They are also propagated to other clusters on the next pull. Imported directory URI catalogs can be updated by uploading a new file. In this case, the changes are propagated at the next pull. Locally configured URIs can be exported in order to be imported on third-party systems that support URI import. ILS-learned URIs and imported URIs are not included in the URI export function. Therefore, to export all URIs of a multicluster deployment, you must export the local URIs at each cluster Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

13 ILS-Based Multicluster URI Routing Process The example in the figure illustrates the URI routing process in an ILS-enabled multicluster deployment. ILS URI syncing is enabled, and URIs are exchanged. When calls the following happens: 1. URI is not found locally using CSS and partitions. 2. Match in ILS table is found; associated route string is emea.cisco.com. 3. SIP route pattern for emea.cisco.com exists and is matched. 4. Call is routed via the associated SIP trunk. ILS URI Syncing: Route String: amer.cisco.com ILS-Learned URIs: -> emea.cisco.com SIP Route Patterns: emea.cisco.com -> amer.cisco.com -> emea.cisco.com Route String: emea.cisco.com ILS-Learned URIs: -> amer.cisco.com SIP Route Patterns: amer.cisco.com 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v The example shows two clusters that are part of an ILS network. URI syncing is enabled, and URIs and route strings have been exchanged. When calls the following URI routing process occurs: Step 1 Step 2 Step 3 The full called URI is looked up in the local URI routing table. The CSS of the calling device is used to determine which partitions of the URI call routing table are considered in the lookup. In the example, the call URI does not exist in the local cluster URI routing table. The full called URI is looked up in the ILS URI routing table. This lookup does not utilize the CSS of the calling device because ILS URIs do not have an assigned partition. When the ILS URI routing table is searched, all patterns are subject to the search. In the example, a match is found. The matched ILS URI entry is associated with route string emea.cisco.com. The route string that is associated with the matched ILS URI entry (emea.cisco.com) is matched against all configured SIP route patterns. In the example, a SIP route pattern for emea.cisco.com is found, and the call is routed according to the SIP route pattern. The CSS of the calling device is used to determine which partitions of the SIP route pattern table are considered in the lookup Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-13

14 ILS Considerations This topic describes what you should consider when implementing ILS. Different sync timers per cluster can result in different convergence times per cluster. - The sync timer can be set differently per cluster. - The maximum sync time is the sum of the sync timers along the sync path. - The sync path has a maximum of three hops. Each cluster that is a member of an ILS network requires a unique cluster ID. ILS relies on TLS for secure information exchange. - Each potential xnode must trust the Tomcat certificate of all other potential xnodes. - Bulk Certificate Management can be used for certificate exchange Cisco and/or its affiliates. All rights reserved. COLLAB90 v Synchronization timers can be configured differently for each cluster. The maximum number of hops in an ILS network is three. The maximum convergence time for a cluster is the sum of the synchronization timers along the synchronization path. By setting all clusters to an identical synchronization timer, an update that occurs in a cluster will be propagated to all clusters after at least three update intervals. If different synchronization timers are used, then clusters that have the same distance to the origin of the update will be updated at different times. This situation makes it difficult to manage such an implementation. Each cluster that is a member of an ILS network must have a unique cluster ID. ILS relies on Transport Layer Security (TLS) for secure information exchange. TLS can be based on certificates or a shared password that is configured at all clusters that participate in the ILS network. In the case of certificates, each potential xnode must trust the Tomcat certificate of all other potential xnodes. The simplest way to exchange the certificates is to use Bulk Certificate Management Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

15 URI Syncing-Related Considerations This section describes URI syncing-related considerations. ILS-learned and imported URIs cannot be assigned a partition. - Local URIs of accessible partitions always have priority. - ILS URIs have priority over lookup of the host portion in SIP route patterns. ILS message exchange is totally independant of SIP route patterns and SIP trunks. - ILS only populates the URI call routing table. - ILS is not used for actual call setups (sending INVITE messages). - SIP route patterns and SIP trunks must be statically configured. Since Cisco Unified Communications Manager version 9, a SIP route pattern can refer to a route list. - The route list allows the configuration of a prioritized list of SIP trunks. - The route list enables the configuration of backup paths Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS-learned and imported URIs cannot be assigned a partition. Locally configured URIs always have priority over ILS URI routing table entries. However, locally configured URIs can be assigned a partition, and therefore not all URIs may be visible to the calling device. ILS URI routing table entries always have priority over a lookup of the host portion against the accessible SIP route patterns. Only if the route string that is associated with a matched ILS URI is not found in any accessible SIP route pattern, the host portion of the URI is matched against the accessible SIP route patterns. Note SIP route patterns can be assigned a partition. SIP route patterns in partitions that are not included in the CSS of the calling devices are ignored. ILS information exchange is totally independent of the signaling message exchange. ILS and URI syncing only populate the ILS URI routing table, but ILS is not used for actual call setups (sending SIP messages such as INVITE). SIP route patterns and SIP trunks must be configured to make the call setup to another cluster work, but they are not required for URI syncing to be successful. Since Cisco Unified Communications version 9, a SIP route pattern can also refer to a route list. By referring to a route list, more than one SIP trunk can be associated with a SIP route pattern to configure backup paths Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-15

16 Using Session Management Edition in an ILS Network This section describes how Cisco Unified Communications Manager Session Management Edition can be used in an ILS network. SIP Route Pattern: * -> SME-ICT SIP Trunk: SME-ICT -> SME Leaf1 ILS Spoke Route String: l1.cisco.com SME ILS Hub Leaf2 ILS Spoke Route String: l2.cisco.com SIP Route Pattern: * -> SME-ICT SIP Trunk: SME-ICT -> SME SIP Route Pattern: * -> SME-ICT SIP Trunk: SME-ICT -> SME Route String: l3.cisco.com Leaf3 ILS Spoke SIP Route Patterns: l1.cisco.com -> L1-ICT l2.cisco.com -> L2-ICT l3.cisco.com -> L3-ICT l4.cisco.com -> L4-ICT SIP Trunks: L1-ICT -> Leaf1 L2-ICT -> Leaf2 L3-ICT -> Leaf3 L4-ICT -> Leaf4 Route String: l4.cisco.com Leaf4 ILS Spoke SIP Route Pattern: * -> SME-ICT SIP Trunk: SME-ICT -> SME 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v When using Cisco Unified Communications Manager Session Management Edition for centralized call processing and monitoring and implementing ILS, the Session Management Edition cluster should be configured as an ILS hub, and all leaf clusters should be configured as ILS spokes. Each ILS spoke is configured with a unique route string. The SIP route pattern configuration at the spokes is very simple. Each cluster has a single SIP route pattern (see the asterisk [*] in the figure), which refers to the only configured SIP trunk that points to the Session Management Edition cluster. At the Session Management Edition cluster, you configure one SIP route pattern and one SIP trunk per ILS spoke cluster. The figure shows an example of an ILS network with a Session Management Edition cluster. Only the spokes have registered endpoints, and the URIs of these endpoints are advertised with a cluster-specific route string Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

17 ILS Implementation This topic describes how to implement ILS. 1. Configure a globally unique enterprise cluster ID. 2. Configure membership of an ILS network: a) Configure the first hub. b) Configure additional hubs. c) Configure spokes. 3. Exchange Tomcat certificates of all potential xnodes (optional). 4. Configure the synchronization interval. 5. Verify remote clusters using the cluster view Cisco and/or its affiliates. All rights reserved. COLLAB90 v This section illustrates the ILS networking configuration procedure, which consists of the following steps: Step 1 Step 2 Step 3 Step 4 Step 5 Configure a globally unique enterprise cluster ID. Configure membership of an ILS network: Configure the first hub. Configure additional hubs. Configure spokes. Exchange Tomcat certificates of all potential xnodes (optional). Configure the synchronization interval. Verify remote clusters using the cluster view Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-17

18 Step 1: Configure the Cluster ID The figure shows the configuration of the cluster ID. The cluster ID must be globally unique when using ILS Cisco and/or its affiliates. All rights reserved. COLLAB90 v The cluster ID is an enterprise parameter and must be set to a unique value when using ILS Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

19 Step 2a: Configure the First Hub Cluster The figure shows the configuration of the first hub cluster. Change the role from Stand Alone Cluster to Hub Cluster. Choose the certificate or password-based ILS authentication. Click the + sign and select cluster members that should run ILS Cisco and/or its affiliates. All rights reserved. COLLAB90 v The first ILS hub cluster is configured as follows: Choose Advanced Features > ILS Configuration. Change the role of the cluster from Stand Alone Cluster to Hub Cluster. Select the ILS authentication method (certificate or password). Note All clusters of an ILS network must share the same authentication method. In the case of password-based authentication, the password must be the same on all clusters. You can click the plus symbol (+) link at the item Server Activation to choose the servers of the cluster on which you want to activate the ILS feature service. After clicking Save, a pop-up window will appear asking for the registration server. Because this is the first ILS hub cluster, leave the registration server field empty Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-19

20 Step 2b: Configure an Additional Hub This section describes how to add an additional ILS hub cluster to the ILS network. After changing the role from Stand Alone Cluster to Hub Cluster, the registration server must be entered. Enter the IP address of the first hub cluster server running ILS Cisco and/or its affiliates. All rights reserved. COLLAB90 v Additional ILS hub clusters are added to an existing ILS network as follows: Choose Advanced Features > ILS Configuration. Change the role of the cluster from Stand Alone Cluster to Hub Cluster. Choose the ILS authentication method (certificate or password). Note The ILS authentication method and the password, if used, must match on all clusters in the same ILS network. You can click the plus symbol (+) link at the item Server Activation to choose the servers of the cluster on which you want to activate the ILS feature service. After clicking Save, a pop-up window will appear asking for the registration server. Enter the IP address of any server of the first ILS hub cluster where the ILS feature service has been activated Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

21 Step 2c: Configure a Spoke This section describes how to add an ILS spoke cluster to the ILS network. After changing the role from Stand Alone Cluster to Spoke Cluster, the registration server must be entered. Enter the IP address of the hub cluster server running ILS Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS spoke clusters are added to an existing ILS network as follows: Choose Advanced Features > ILS Configuration. Change the role of the cluster from Stand Alone Cluster to Hub Cluster. Choose the ILS authentication method (certificate or password). You can click the plus symbol (+) link at the item Server Activation to choose the servers of the cluster on which you want to activate the ILS feature service. After clicking Save, a pop-up window will appear asking for the registration server. Enter the IP address of any server of the ILS cluster that acts as a hub to the currently configured spoke cluster where the ILS feature service has been activated Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-21

22 Step 3: Exchange Tomcat Certificates This section describes how to exchange Tomcat certificates between all potential xnodes. The simplest way to exchange Tomcat certificates is by using Bulk Certificate Management. It is only required if certificate-based ILS authentication is chosen Cisco and/or its affiliates. All rights reserved. COLLAB90 v When the ILS authentication method must be set to Use TLS Certificates, then you must export the Tomcat certificate of each server that could be designated as an xnode. Then you must import the Tomcat certificates of all other potential xnodes at each potential xnode. These certificates can be imported, one by one, by using Cisco Unified Operating System Administration or by using Bulk Certificate Management Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

23 Step 4: Configure the Synchronization Interval This section describes how to set the synchronization interval at an ILS cluster. If desired, change the synchronization interval (default: 10 minutes) Cisco and/or its affiliates. All rights reserved. COLLAB90 v At the ILS Configuration page, you can change the synchronization interval. The default is 10 minutes. Note The synchronization interval is configured per cluster. It is recommended that you use the same value at all clusters of an ILS network Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-23

24 Step 5: Verify ILS Networking Using Cluster View This section shows how to verify ILS networking configuration and operation. Select the remote cluster Cisco and/or its affiliates. All rights reserved. COLLAB90 v You can view remote clusters and their services by choosing Advanced Features > Cluster View. The cluster view lists all remote clusters that provide services to the local cluster. If you click an entry, the services that are available at this cluster are listed. Note The cluster view was utilized for EMCC only before Cisco Unified Communications Manager version 9. Beginning with version 9, it also shows services such as link bandwidth management, UDS, and ILS (if enabled) Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

25 ILS-Based URI Routing Configuration Procedure This section describes how to configure ILS-based URI routing. Configure URI syncing: 1. Enable URI syncing via ILS. 2. Configure a globally unique route string. Configure URI routing: 1. Configure route groups and route lists (optional). 2. Configure SIP profiles. 3. Configure SIP trunks. 4. Configure SIP route patterns Cisco and/or its affiliates. All rights reserved. COLLAB90 v In order to implement ILS-based URI routing, you must join the participating clusters into an ILS network. The related configuration steps were shown at the beginning of this topic. In addition to a functioning ILS network, you must enable URI syncing and configure URI routing including SIP trunks, route lists (optional), and SIP route patterns. The URI routing configuration procedure is the same for a deployment with ILS or without ILS. In an ILS deployment, you must make sure that the SIP route patterns match the ILS route strings and not the host portion of the dialed URIs. Note The configuration of SIP route patterns and SIP trunks was explained in an earlier lesson of this training. The URI syncing configuration procedure consists of two steps: Step 1 Step 2 Enable URI syncing via ILS. Configure the route string to be associated with the local URIs when they are advertised by ILS Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-25

26 Enable URI Syncing and Configure the Route String The figure shows how to enable URI syncing and how to configure the route string that will be advertised along with the locally configured URIs. Enable URI syncing. Set the route string Cisco and/or its affiliates. All rights reserved. COLLAB90 v Perform the following steps to enable URI syncing: Choose Call Routing > Intercluster Directory URI > Intercluster Directory URI Configuration. Check the Exchange Directory URI Catalogs with Remote Clusters check box. Enter the route string to be used by the currently configured cluster Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

27 ILS Monitoring This topic describes how to monitor ILS operation. Command utils ils findroutestring uri utils ils lookup uri utils ils findxnode utils ils showpeerinfo Description Displays all route strings that are associated with the specified URI. Displays the route string that is used to route a call to the specified URI. Displays the current xnode of the local cluster. Displays information such as the route string, enterprise cluster ID, and role (hub versus spoke) about ILS peers Cisco and/or its affiliates. All rights reserved. COLLAB90 v ILS operation can be monitored using the CLI of any server where the ILS feature service has been activated. The table shows the ILS-related commands and their functions. In a correctly configured ILS deployment, there should be only one route string for a given URI. In other words, a URI should not be present in more than one cluster. If there were multiple route strings that are associated with the specified URI, the first command would display all of them. Therefore, the first command is ideal to troubleshoot routing issues that are caused by duplicate ILS URIs. The second command displays the route string that is used to route a call to the specified URI. Even if multiple route strings were associated with the specified URI, the second command would only display one route string. The third command identifies the xnode of the cluster. Knowing which server has been designated as the xnode is important when troubleshooting of the ILS exchange with remote clusters is required, because only the xnode is actively participating in this exchange. Typically, you would use the third command to identify the xnode and then start tracing ILS on this cluster node. The last command shows the details of ILS peers. This command includes the route string, the enterprise cluster ID, and the role of a specified ILS peer Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-27

28 ILS Alarms This section lists and describes the alarms that are available to monitor ILS. The following ILS-related alarms have been added to Cisco Unified Communications Manager version 9: TLSAuthenticationFailed TCPInUse TCPListenError ILSHubClusterUnreachable ILSRemoteHostUnresponsive ILSProtocolVersion ILSUnacceptableConnectionAttempt ILSDuplicateURIOnLookup ILSDuplicateURIOnReplication 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v These ILS-related alarms have been added to Cisco Unified Communications Manager version 9: TLSAuthenticationFailed: This alarm indicates that a certificate was received that has not been installed locally, and the fully qualified domain name (FQDN) is not a known remote cluster. This case fails the authentication requirements. TCPInUse: This alarm returns all of the route strings that are associated with the URI, including the local cluster. It can be used to troubleshoot problems involving duplicate URIs. TCPListenError: This alarm is raised if the ILS port (7501 by default) is already in use when ILS starts up and attempts to bind to it. ILSHubClusterUnreachable: This alarm is raised when a connection to a hub cluster cannot be established. This alarm will be raised on the xnode. ILSRemoteHostUnresponsive: This alarm is raised when a connection to another cluster is unexpectedly lost. This alarm will be raised on the xnode. ILSProtocolVersion: This alarm is raised when two clusters are running incompatible versions of ILS. They will not be able to communicate and share cluster information or URIs. ILSUnacceptableConnectionAttempt: This alarm is raised in the following scenarios: A new cluster attempted to join, but the local cluster is not configured as a hub. An unknown cluster has attempted to connect without first registering to the ILS network. Data corruption in the network packet can also cause this alarm Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

29 ILSDuplicateURIOnLookup: This alarm is raised when a URI lookup results in more than one routing string. This alarm is raised on any node that runs ILS. ILSDuplicateURIOnReplication: This alarm is raised when a URI from a remote cluster is replicated in and matches a local URI. Note Because of performance effects, the check for a match on a local URI is not performed by default, and therefore the alarm will never activate. If needed, Cisco Technical Assistance Center (TAC) will provide you with instructions on how to enable the check. This alarm is operational only when this check is enabled Cisco Systems, Inc. Implementing Intercluster Lookup Service 11-29

30 Summary This topic summarizes the key points that were discussed in this lesson. ILS is a new feature that was introduced in Cisco Unified Communications Manager version 9 that enables intercluster URI routing with overlapping URI host portions. All clusters that should exchange ILS information must be joined into the same ILS network. URI syncing allows clusters that are part of an ILS network to exchange their local URIs along with a cluster-specific route string. ILS-learned URIs are only considered if the called URI is not found in the local database. ILS implementation steps include ILS network and URI syncing configuration tasks. ILS monitoring options include alarms and CLI commands Cisco and/or its affiliates. All rights reserved. COLLAB90 v References This lesson described how ILS works and how it is implemented. The lesson started with an overview of ILS and then described the functions of ILS networking and URI syncing. The lesson then explained what needs to be considered when implementing ILS and how to configure ILS. Finally, the lesson described how to monitor ILS operation. For additional information, refer to this resource: Cisco Systems, Inc., Cisco Unified Communications Manager Features and Services Guide, Release 9.1(1), section Intercluster Lookup Service at: K_C3E0EFA0_00_cucm-features-services-guide-91_chapter_ html Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

31 Appendix 12 Implementing Advanced SIP Solutions Overview Objectives This lesson provides an overview of advanced Session Initiation Protocol (SIP) implementations. It describes SIP features and functions, endpoint registration options, and includes considerations when implementing SIP with other vendors. The end of this lesson focuses on troubleshooting advanced SIP implementations. Upon completing this lesson, you will be able to implement advanced SIP solutions, including multivendor environments. This ability includes being able to meet these objectives: Describe advanced SIP implementation List and describe the SIP features and functions Describe the options for SIP endpoint registration in Cisco Unified Communications Manager version 9 Troubleshoot advanced SIP solutions

32 Overview of Advanced SIP Implementations This topic provides an overview of SIP. SIP is an IETF standard. It creates, modifies, and terminates multimedia sessions with one or more participants It is supported on all Cisco collaboration endpoints. It is a peer-to-peer architecture: - UAC initiates SIP requests. - UAS returns SIP responses. - Phones, gateways, and Cisco call control devices can be UAs. SIP uses ASCII text-based messages 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v The IETF developed SIP. SIP is a peer-to-peer, common standard protocol that is based on the logic of the World Wide Web and very simple to implement. SIP is designed to address the functions of signaling and session management. It is widely used with proxy servers, gateways, phones, and Internet telephony service providers (ITSPs). Through invitations, SIP initiates sessions or invites participants into established sessions. Descriptions of these sessions are advertised by any one of several means, including the Session Announcement Protocol (SAP). SAP incorporates a session description according to the Session Description Protocol (SDP). The user agent (UA) is a combination of the user agent client (UAC) and the user agent server (UAS) that initiates and receives calls. A UAC initiates a SIP request. A UAS, which is a server application, contacts the user when it receives a SIP request. The UAS then responds on behalf of the user. Cisco Unified Communications Manager can act as both a server and a client (a back-to-back user agent) Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

33 SIP Trunk Overview This subtopic provides an overview of SIP trunks. SIP trunks provide connectivity to other SIP devices: - Gateways - Cisco Unified CM Session Management Edition - SIP proxies - Other Cisco Unified CM clusters - Third-party vendors SIP trunk and call routing enhancements: - Can run on all Cisco Unified CM nodes - Up to 16 destination IP addresses per trunk - SIP early offer support for voice and video calls (insert MTP, if needed) - Audio codec preference (Accept Audio Codec Preference in Received Offer) - SIP trunk normalization and transparency - Supports the use of route lists on all Cisco Unified CM nodes 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v SIP trunks provide connectivity to other SIP devices such as gateways, Cisco Unified Communications Manager Session Management Edition, SIP proxies, Cisco Unified Communications applications, and other Cisco Unified Communications Manager clusters. Today, SIP is arguably the most commonly chosen protocol when connecting to service providers and Cisco Unified Communications applications. Cisco Unified Communications Manager provides the following SIP trunk and call routing capabilities: Can run on all Cisco Unified Communications Manager nodes Up to 16 destination IP addresses per trunk SIP early offer support for voice and video calls with media termination point (MTP) insertion only if needed Audio coder-decoder (codec) preference lists and option to accept audio codec preference in received offer SIP trunk normalization and transparency Supports the use of route lists on all Cisco Unified Communications Manager nodes The SIP trunk features that are available in the current release of Cisco Unified Communications Manager make SIP the preferred choice for new and existing trunk connections. The ability to run on all Cisco Unified Communication Manager nodes and to manage up to 16 destination IP addresses improves outbound call distribution from Cisco Unified Communications Manager clusters and reduces the number of SIP trunks that are required between clusters and devices. SIP early offer support for voice and video calls (insert MTP, if needed) can reduce or eliminate the need to use MTPs and allows voice, video, and encrypted calls to be made over SIP early offer trunks Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-3

34 Review of SIP Features and Functions This topic describes SIP features and functions. Determines the location of the destination endpoint Determines the media capabilities of the destination endpoint - SDP establishes the best common service level Determines the availability of the destination endpoint and informs the reason of termination - Not reachable - Busy - No answer Establishes a session between participant endpoints Manages the transfer and termination of calls 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v SIP supports five key functions when establishing a communication: SIP determines the location of the destination endpoint: Address resolution Name mapping Call redirection SIP determines the media capabilities of the destination endpoint. SDP message codec negotiation is used to determine the best codec that is common between endpoints. SIP determines the availability of the destination endpoint. SIP determines whether the called party is already connected to a call or did not answer. If the call cannot be completed, SIP returns a message that indicates the reason why the target endpoint was unavailable. SIP establishes a session between participant endpoints. When the call can be completed, SIP establishes a session between the endpoints. SIP manages the transfer and termination of calls. During a call, SIP establishes a session between the transferee and a new endpoint. SIP tears down the session between the endpoints when the call is terminated Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

35 SIP Call Setup This topic describes how a basic SIP call is set up and terminated in a Cisco Unified Communications Manager deployment. 1. INVITE (SDP) Trying 2. INVITE (SDP) Ringing Ringing OK OK 6. ACK 6. ACK AUDIO VIDEO 7. Media (UTP) 8. BYE 8. BYE OK OK 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v This figure depicts the call setup and teardown between two SIP endpoints. SIP call setup proceeds as follows: Step 1 The originating endpoint sends an invitation (INVITE) to the Cisco Unified Communications Manager. The message includes the SDP description of the supported media parameters. Step 2 Step 3 Step 4 Step 5 Step 6 Step 7 Step 8 Step 9 The Cisco Unified Communications Manager forwards this invitation to the targeting endpoint. The originating endpoint receives a 100 Trying message from Cisco Unified Communications Manager, which indicates that the INVITE was sent to the targeting endpoint. The terminating endpoint sends the ringing signal to the originating endpoint. The call is taken off-hook. If the targeting endpoint determines that the call parameters are acceptable, it responds to the originator using the 200 OK message. The originating endpoint issues an acknowledgment (ACK) to the targeting endpoint. A Real-Time Transport Protocol (RTP) or UDP session is established between the endpoints. The originating endpoint terminates the call, and the BYE message is sent to the targeting endpoint via Cisco Unified Communications Manager. The remote endpoint confirms the BYE message with 200 OK messages, which is also exchanged via Cisco Unified Communications Manager Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-5

36 Session Description Protocol SDP is part of a SIP message. SDP describes session parameters in a SIP message. SDP contains the following: - Type of media (audio and media) - Transport protocol (RTP or UDP or IP and H.230) - Media format (codecs) A list of supported media formats is offered: - All listed formats may be used in the session. SIP exchanges codecs at different stages in call setup: - Delayed offer: 200 OK and ACK - Early offer: Invite and 200 OK - Early media: 183 Session Progress, 180 Ringing, Pre-Ack 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v SDP is an IETF-based format for describing streaming media initialization parameters. SDP is intended for describing multimedia communication sessions for the purposes of session announcement, session invitation, and parameter negotiation. SDP does not deliver media itself but is used for negotiation between endpoints of media type, format, and all associated properties. SDP is designed to be extensible to support new media types and formats. SIP uses SDP to negotiate the type of media (audio and video), the transport protocol (RTP), and the media format (audio and video codecs). The originating endpoint provides a selection list of supported codecs, which is ordered by highest priority. The targeting endpoint chooses an offer that matches its capabilities. SIP uses the Offer-Answer model to establish SIP sessions, and SIP can exchange codecs at different stages in the call setup Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

37 Example: Audio, RTP/49174, G.711 mulaw Video, RTP/49170, H261 v=0 o=bob IN IP4 host1.cisco.com s=example c=in IP t=0 0 m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 m=video RTP/AVP 31 a=rtpmap:31 H261/90000 Field Description Version v=0 Origin o=<username> <session id> <version> <network type> <address type> <address> Session s=<session name> name Connection data Times Media Audio Video Profile (AVP) codes c=<network type> <address type> <connection address> t=<start time> <stop time> m=<media> <port> <transport> <media format list> 0: G.711 mu-law 8: G.711 a-law 3: GSM codec 18: G : H Cisco and/or its affiliates. All rights reserved. COLLAB90 v This figure is an example of SDP. The table explains the parameters that are used in the example: Version: This parameter is the protocol version. Origin: This parameter describes the originator of the message. Session name: This parameter is a mandatory name with at least one UTF-8-encoded character. Times: This parameter includes the start and end time of a session. Connection data: This parameter provides the parameters for media endpoint termination: network type (IN is defined as Internet), address type (IPv4 or IPv6), and the connection address (IP address). Media: This parameter specifies the media type (audio or video), the UDP transport port, and all offered media formats. Examples of Audio Video Profile (AVP) codes are 0 (G.711 mu-law), 8 (G.711 a-law), 3 (GSM codec), 18 (G.729), and 31 (H.261). Offered codecs are ordered according to the priority (from most preferred to least preferred) Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-7

38 Delayed Offer Delayed offer is one method of negotiating media. 1. INVITE Trying 2. INVITE Ringing Ringing OK (SDP: Media Offer) OK (SDP: Media Offer) 6. ACK (SDP: Media Answer) 6. ACK (SDP: Media Answer) AUDIO VIDEO Media (UTP) 7. BYE 7. BYE OK OK 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v In a delayed offer, the session initiator does not send its media capabilities in the INVITE message but waits for the called device to send its capabilities first. The codec that is used in this scenario is determined by the targeting endpoint. The originating endpoint receives the codec priority list from the targeting endpoint and compares the list with its own codec priority list, which is configured in regional settings. The codec priority list is compared from the first (prioritized) codec to the last (least prioritized) codec. The first prioritized codec that is supported by both endpoints is chosen to negotiate the call. Note Cisco gateways support the delayed offer as well, but an originating gateway defaults to the early offer Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

39 Early Offer The INVITE message contains the SDP media offer when the call is initialized. 1. INVITE (SDP: Media Offer) Trying 2. INVITE (SDP: Media Offer) Ringing Ringing OK (SDP: Media Answer) OK (SDP: Media Answer) 6. ACK 6. ACK AUDIO VIDEO Media (UTP) 7. BYE 7. BYE OK OK 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v When using early offer, the session initiator sends its capabilities (including the supported codecs) in the SDP, which is contained in the initial INVITE message. This method allows the called device to choose its preferred codec for the session. Early offer is the default method that is used by Cisco Unified Border Element when acting as the originating gateway. To enable early offer on Cisco Unified Communications Manager, the SIP profile must be adjusted appropriately Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-9

40 Early Media Sending of media can be allowed before the call is accepted. 183 Session Progress Option 1. INVITE Trying 2. INVITE Session Progress (SDP: MO) Session Progress (SDP: Media Offer) 5. Pre-ACK (SDP: Media Response) 5. Pre-ACK (SDP: Media Response) Ringing Ringing AUDIO VIDEO Media (UTP) OK OK 8. ACK 8. ACK 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v Early media allows the sending of media from the called party or an application server to the caller, even before the call is accepted. The most common reasons for using early media include the following: The called device might want to establish an early media RTP path to reduce the effects of audio cut-through delay (clipping) for calls that experience long signaling delays or to provide a network-based voice message to the caller. The calling device might want to establish an early media RTP path to access a dual tone multifrequency (DTMF) or voice-driven interactive voice response (IVR) system. Cisco Unified Communications Manager supports Provisional Reliable Acknowledgement (PRACK)-based early media for both early offer and delayed offer calls. Early two-way media cut-through can be achieved by enabling PRACK on each SIP Cisco Unified Communications system. PRACK allows the SIP offer and answer to be sent reliably in provisional responses (for example, 1XX responses), thus reducing the number of messages that need to be exchanged before two-way media can be established Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

41 180 Ringing Option 1. INVITE Trying 2. INVITE Ringing (SDP: Media Offer) Ringing (SDP: Media Offer) 5. Pre-ACK (SDP: Media Response) 5. Pre-ACK (SDP: Media Response) AUDIO VIDEO Media (UTP) OK OK 8. ACK 8. ACK 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v To facilitate early media with delayed offer, the IETF draft allows the use of messages other than the 183 Session Progress response. Some implementations use the 180 Ringing response to send the initial SDP media offer. The 180 Ringing message is a provisional or informational response that is used to indicate that the INVITE message has been received by the user agent and that alerting is taking place Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-11

42 Media Termination Point If DTMF between endpoints is inconsistent, an MTP can be dynamically located. An MTP can be configured to be used on the following at all times: - Lines - Trunks By default, it is not used by any SIP device. It is dynamically located when the DTMF method between connected devices is not compatible. SIP IP Phone PRI Gateway TDM PSTN RTP Stream RTP Stream MTP 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v MTPs are used by Cisco Unified Communications Manager to accomplish the following goals: Deliver a SIP early offer over SIP trunks Address DTMF transport mismatches Act as an RSVP agent Act as a trusted relay point (TRP) Provide conversion between IPv4 and IPv6 for voice RTP streams Either of the following methods can be used to enable early offer on SIP trunks: Check the MTP Required check box on the SIP trunk. In this case, an MTP is used for every outbound call, and only voice calls are supported. You can choose MTP Preferred Originating Codec from the drop-down list under SIP trunk configuration. Check the Early Offer Support for Voice and Video Calls (Insert MTP If Needed) check box on the SIP profile that is associated with the SIP trunk. With this method, an MTP is inserted only if the calling device or trunk cannot send all of the information about its media capabilities in the initial SIP invite (for example, an inbound call to Cisco Unified Communications Manager from a SIP delayed offer or H.323 Slow Start trunk). In this case, when an MTP is used, additional voice codecs can be supported in the initial call setup by using the pass-through codec of the MTP. Once established, this audio call can be escalated to support video and encryption if the media of the call is renegotiated (for example, after hold or resume). When an MTP is not needed, all calls support voice, video, and encrypted media Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

43 SIP Endpoint Registration Options This topic describes feature support in Cisco SIP IP phones. When using SIP on a Cisco IP phone, only basic features are supported on Type-A IP phones: - Cisco IP Phones 7905, 7912, 7940, and 7960 have end-of-life status and can no longer be purchased. On newer IP phones, most of the features are supported when using SIP: - Cisco Unified IP Phones 794(25), 796(25), and Cisco Unified IP phones with SIP-only support: - Cisco Unified IP Phones 8961, 9951, and Third-party IP phones support limited features when registered to Cisco Unified Communications Manager. Blended addressing is not supported with Type-A IP phones Cisco and/or its affiliates. All rights reserved. COLLAB90 v Type-A Cisco IP phones support basic features when using SIP firmware. Supported features are Answer Release, Auto Answer, Call Forward, Call Waiting, Caller ID, Conference, Do Not Disturb, Help System, Hold/Resume, Mute, Redial, Speed Dialing, Transfer, and Voice Mail. Other features are not supported or have only limited support. Note Cisco IP Phones 7905, 7912, 7940, and 7960 have end-of-life status and can no longer be purchased. On newer Cisco IP phones, there are very few features unsupported when using SIP. Blended addressing is supported by these IP phones, and it is enabled by default on Cisco Unified Communications Manager. Note Blended addressing will insert both the directory URI and the directory number of the sending party in outgoing SIP invites. Third-party SIP phones register with Cisco Unified Communications Manager but do not use a MAC address-based device ID. User-based registration must be used to identify a registering third-party SIP phone. Configuration for SIP registration needs to be done on Cisco Unified Communications Manager and on third-party SIP phones Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-13

44 Third-Party SIP Endpoints This subtopic describes SIP features for third-party SIP endpoints. Third-party SIP IP phones support only a few features compared with Cisco SIP IP phones. The features that are not supported are the following: - MAC address registration - Phone button templates - Softkey templates - Telephony features and applications: Cisco Unified Video Advantage Cisco Unified Communications Manager Assistant Cisco IP Phone Services Call Pickup Barge Cisco Unified Presence 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v The limitations of third-party SIP endpoints include, but are not limited to, the following: MAC address-based registration. SIP phones must be configured and associated with the user information that is used for phone registration (user ID and extension number) instead of a MAC address-based device ID. There is no support for phone templates and softkey templates. The user interface depends on the SIP product that is used. Telephony features and applications such as the following are not supported: Cisco IP Phone Services Cisco Unified Communications Manager Assistant Cisco Unified Video Advantage Call Pickup Barge Cisco Unified Presence Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

45 Cisco SIP IP Phones SIP-only Cisco IP phones support a large number of features. Cisco Unified IP Phones 8961, 9951, and 9971 support only the SIP protocol. Features that are supported when registering these IP phones to Cisco Unified CM include the following: - Autoregistration - MAC address registration - Phone button templates - Softkey templates - Blended addressing - Video - URI dialing 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v Cisco Unified IP Phones 8961, 9951, and 9971 are SIP-only IP phones that can register to Cisco Unified Communications Manager with the MAC address. For more information about the Cisco Unified IP Phone 8900 and 9900 Series, refer to the following data sheets and product documentation: Cisco Unified IP Phone 8900 Series: Cisco Unified IP Phone 9900 Series: Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-15

46 Cisco SIP IP Phone Registration Prerequisites It is mandatory to choose a phone security profile and a SIP profile. When registering a Cisco SIP IP phone, the following can be preconfigured: - Phone security profile - SIP profile Depending on the type of SIP IP phone, the system parameters must be set. Phone Security Profile SIP Profile (SIP Phones Only) 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v Cisco Unified Communications Manager Administration groups security-related settings in the SIP phone security profile, which works like a template. You will configure the settings in the profile and then apply the profile to the SIP phone. After you apply the profile to the phone, it uses the settings that you configured in the SIP Phone Security Profile window. If you want to do so, you can use the same profile for many SIP phones, which eliminates having to configure the same set of parameters multiple times. The SIP Phone Security Profile window includes security-related settings such as Device Security Mode, Authentication Mode (for CAPF), Key Size (for CAPF), Enable Digest Authentication, Nonce Validity Time, Transport Type, and SIP Phone Port. Note All SIP phones require that you apply a security profile. A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call-pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

47 Phone Security Profile Configuration The phone security profile must be configured to enable authentication and encryption. Choose System > Security > Phone Security Profile. Find the security profile that contains the IP phone model number in the name field. Click Cisco 9951 Standard SIP Non-Secure Profile. To change the settings of an inbuilt phone security profile, you need to make a copy first and give it a different name Cisco and/or its affiliates. All rights reserved. COLLAB90 v Perform the following steps to configure the phone security profile: Step 1 Step 2 Step 3 Choose System > Security > Phone Security Profile. Find the security profile that contains the IP phone model number in the name field. In this example, click Cisco 9951 Standard SIP Non-Secure Profile to create a phone security profile for the Cisco Unified IP Phone 9951 phone model. Note Step 4 Depending on the chosen phone model, the phone security profile will only include the configuration parameter that is applicable to the chosen phone. Click Copy and name the newly created phone security profile Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-17

48 Device Security Modes: Non Secure Authenticated Encrypted Transport Types in Non Secure Mode: TCP UDP TCP+UDP Transport Types in Secure Mode: TLS To enable Digest Authentication, check this box. Enable IP phone configuration encryption by checking this box Cisco and/or its affiliates. All rights reserved. COLLAB90 v The Phone Security Profile Information provides security configuration options as follows: Device Security Mode Non Secure: This option provides no security features except image, file, and device authentication for the phone. A TCP connection opens to Cisco Unified Communications Manager. Authenticated: Cisco Unified Communications Manager provides integrity and authentication for the phone. Packets are protected against modifications but not against eavesdropping. Encrypted: Cisco Unified Communications Manager provides integrity, authentication, and encryption for the phone. Packets are protected against modifications and against eavesdropping (Secure RTP [SRTP]). Transport Type: When the Device Security Mode is Non Secure, choose one of the following options: TCP: Choose TCP to ensure that packets are received in the exact order in which they are sent. This protocol ensures that no packets are dropped, but the protocol does not provide any security. UDP: Choose UDP to ensure that packets are received quickly. This protocol, which can drop packets, does not ensure that packets are received in the order in which they are sent. This protocol does not provide any security. TCP + UDP: Choose this option if you want to use a combination of TCP and UDP. This option does not provide any security. Transport Type: When Device Security Mode is Authenticated or Encrypted, Transport Layer Security (TLS) can be chosen. TLS provides signaling integrity, device authentication, and signaling encryption (encrypted mode only) for SIP phones Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

49 Enable Digest Authentication: If you check this check box, Cisco Unified Communications Manager challenges all SIP requests from the phone. Digest authentication does not provide device authentication, integrity, or confidentiality. TFTP Encrypted Config: When this check box is checked, Cisco Unified Communications Manager encrypts phone downloads from the TFTP server Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-19

50 SIP Profile Configuration Configure a SIP profile in Cisco Unified Communications Manager Administration. Choose Device > Device Settings > SIP Profile. Click Find, and make a copy of the standard SIP Profile Cisco and/or its affiliates. All rights reserved. COLLAB90 v The SIP profile offers many options to be configured and applied to a SIP phone or SIP trunk. The example shows the configuration of the following options: Default MTP Telephony Event Payload Type: This field specifies the default payload type for an RFC 2833 telephony event. The default value specifies 101 with a range from 96 to 127. If the ITSP is using a different DTMF payload type, you can change it here. Dial String Interpretation: Cisco Unified Communications Manager uses the Dial String Interpretation policy to determine if the SIP identity header is a directory number or directory URI. Because directory numbers and directory URIs are saved in different database lookup tables, it is important to choose the right option that clearly determines the directory number or directory URI. Redirect by Application: Checking this check box and configuring this SIP profile on the SIP trunk allows Multiway functionality to work properly when triggered from remotely registered video telepresence devices. Note Multiway conferencing enables video endpoint users to introduce a third party into an existing call. Disable Early Media on 180: If you do not receive ringback, the device to which you are connecting may include SDP in the 180 response but is not sending any media before the 200 OK response. In this case, check this check box to play a local ringback tone on the calling phone and connect the media upon receipt of the 200 OK response Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

51 When the SIP device is a tablet device, the following parameters should be changed: - Timer Register Delta (seconds): 60 - Timer Register Expires (seconds): Timer Keep Alive Expires (seconds): Timer Subscribe Expires (seconds): Cisco and/or its affiliates. All rights reserved. COLLAB90 v Example of a SIP Profile for Cisco Jabber Running on Tablets The example shows a SIP profile that allows Cisco Jabber running on tablets to stay connected to Cisco Unified Communications Manager while the application is running in the background. To configure the SIP profile, perform the following steps: Step 1 Step 2 Step 3 Step 4 Choose Device > Device Settings > SIP Profile. Create a SIP profile or copy an existing SIP profile. You can name the profile Standard Tablet SIP Profile. In the Parameters Used in Phone section, enter these values: Timer Register Delta (seconds): 60 (default: 5) Timer Register Expires (seconds): 660 (default: 3600) Timer Keep Alive Expires (seconds): 660 (default: 120) Timer Subscribe Expires (seconds): 660 (default: 120) Click Save Cisco Systems, Inc. Implementing Advanced SIP Solutions 12-21

52 SIP Multivendor Implementation Considerations This topic describes how to implement Cisco Unified Communications Manager with multiple vendors via SIP. Internet Telephony Service Provider SIP Trunk SIP Video TelePresence Cisco VCS SIP IP Phone SIP Video Phone SIP Video TelePresence SIP Proxy Third-Party Class 4/5 Switch TelePresence Endpoint with Twin Data Display 2013 Cisco and/or its affiliates. All rights reserved. COLLAB90 v Cisco Unified Communications Manager supports connecting to numerous vendors and call processing servers via SIP trunks. It supports aggregating different trunk connections and interconnecting them. Cisco Unified Communications Manager can provide redundancy and failover with SIP trunking as well. Cisco Unified Communications Manager only accepts calls from the SIP device whose IP address matches the destination address of the configured SIP trunk. In addition, the port on which the SIP message arrives must match the one that is configured on the SIP trunk Appendix Collaboration 9.0 Features 2013 Cisco Systems, Inc.

Intercluster Lookup Service

Intercluster Lookup Service When the (ILS) is configured on multiple clusters, ILS updates Cisco Unified Communications Manager with the current status of remote clusters in the ILS network. The ILS cluster discovery service allows

More information

URI Dialing. Set Up URI Dialing

URI Dialing. Set Up URI Dialing Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an

More information

IP Phone Presence Setup

IP Phone Presence Setup Static Route Configuration on IM and Presence Service, page 1 Presence Gateway Configuration on IM and Presence Service, page 6 Configure SIP Publish Trunk on IM and Presence Service, page 7 Configure

More information

Introducing Cisco Voice and Unified Communications Administration Volume 1

Introducing Cisco Voice and Unified Communications Administration Volume 1 Introducing Cisco Voice and Unified Communications Administration Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your

More information

Acano solution. Third Party Call Control Guide. March 2015 76-1055-01-E

Acano solution. Third Party Call Control Guide. March 2015 76-1055-01-E Acano solution Third Party Call Control Guide March 2015 76-1055-01-E Contents Contents 1 Introduction... 3 1.1 How to Use this Guide... 3 1.1.1 Commands... 4 2 Example of Configuring a SIP Trunk to CUCM...

More information

Enabling Users for Lync services

Enabling Users for Lync services Enabling Users for Lync services 1) Login to collaborate.widevoice Server as admin user 2) Open Lync Server control Panel as Run As Administrator 3) Click on Users option and click Enable Users option

More information

IP Office Technical Tip

IP Office Technical Tip IP Office Technical Tip Tip no: 188 Release Date: September 27, 2007 Region: GLOBAL Verifying IP Office SIP Trunk Operation IP Office back-to-back SIP Line testing IP Office Release 4.0 supports SIP trunking.

More information

Cisco Unified Communications Manager with Cisco Expressway (SIP Trunk)

Cisco Unified Communications Manager with Cisco Expressway (SIP Trunk) Cisco Unified Communications Manager with Cisco Expressway (SIP Trunk) Deployment Guide Cisco Expressway X8.2 Unified CM 8.6.x, 9.x January 2015 Contents Introduction 4 Deployment scenario 4 Configuring

More information

SBC 1000 / SBC 2000 Series Configuration Guide (For Microsoft Lync Server 2013)

SBC 1000 / SBC 2000 Series Configuration Guide (For Microsoft Lync Server 2013) Configuration Guide SBC 1000 / SBC 2000 Series Configuration Guide (For Microsoft Lync Server 2013) For use with AT&T s IP Flexible Reach Enhanced Features Service on MIS, MPLS PNT or AT&T VPN Disclaimers

More information

Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led

Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the

More information

CISCO UNIFIED COMMUNICATIONS MANAGER

CISCO UNIFIED COMMUNICATIONS MANAGER CISCO UNIFIED COMMUNICATIONS MANAGER V10 UPDATED TOPICS LOCAL ROUTE GROUP ENHANCEMENT Multiple Local Route Groups can be associated with Route Groups for Emergency Dialing. In releases 8 and 9, administrators

More information

Integrating Avaya Aura Presence Services with Microsoft OCS

Integrating Avaya Aura Presence Services with Microsoft OCS Integrating Avaya Aura Presence Services with Microsoft OCS 6.1 Service Pack 5 December 2012 Contents Chapter 1: Introduction... 5 Overview - OCS/Lync integration... 5 The Presence Services server and

More information

Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led

Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 2 (CIPTV2) v1.0 is a five-day course that prepares the

More information

Unified Communications Mobile and Remote Access via Cisco VCS

Unified Communications Mobile and Remote Access via Cisco VCS Unified Communications Mobile and Remote Access via Cisco VCS Deployment Guide Cisco VCS X8.2 Cisco Unified CM 9.1(2)SU1 or later January 2015 Contents Mobile and remote access overview 4 Jabber client

More information

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Abstract These Application

More information

"Charting the Course... Implementing Cisco IP Telephony & Video, Part 1 v1.0 ( CIPTV1 ) Course Summary

Charting the Course... Implementing Cisco IP Telephony & Video, Part 1 v1.0 ( CIPTV1 ) Course Summary Description Course Summary Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the learner for implementing a Cisco Collaboration solution at a single-site

More information

Cisco EXAM - 300-075. Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product. http://www.examskey.com/300-075.

Cisco EXAM - 300-075. Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product. http://www.examskey.com/300-075. Cisco EXAM - 300-075 Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product http://www.examskey.com/300-075.html Examskey Cisco 300-075 exam demo product is here for you to test the

More information

IMPLEMENTING CISCO IP TELEPHONY & VIDEO, PART 1 V1.0 (CIPTV1)

IMPLEMENTING CISCO IP TELEPHONY & VIDEO, PART 1 V1.0 (CIPTV1) IMPLEMENTING CISCO IP TELEPHONY & VIDEO, PART 1 V1.0 (CIPTV1) COURSE OVERVIEW: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the learner for implementing

More information

TROUBLESHOOTING CISCO IP TELEPHONY & VIDEO V1.0 (CTCOLLAB)

TROUBLESHOOTING CISCO IP TELEPHONY & VIDEO V1.0 (CTCOLLAB) TROUBLESHOOTING CISCO IP TELEPHONY & VIDEO V1.0 (CTCOLLAB) COURSE OVERVIEW: Troubleshooting Cisco IP Telephony & Video (CTCOLLAB) v1.0 is a five-day course that prepares the learner for troubleshooting

More information

Configuration Note for Jeron Provider 790 and Cisco CallManager

Configuration Note for Jeron Provider 790 and Cisco CallManager Configuration Note for Jeron Provider 790 and Cisco CallManager 1. Jeron Provider 790 Setup 1.1 Configure the SIP Server Connectivity Set Brekeke SIP Server's IP address in the [SIP Server IP] field and

More information

IM and Presence Service Network Setup

IM and Presence Service Network Setup Configuration changes and service restart notifications, page 1 Domain Value Configuration, page 2 Routing Information Configuration on IM and Presence Service, page 3 Configure Proxy Server Settings,

More information

NAT TCP SIP ALG Support

NAT TCP SIP ALG Support The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the

More information

Unified Communications Mobile and Remote Access via Cisco Expressway

Unified Communications Mobile and Remote Access via Cisco Expressway Unified Communications Mobile and Remote Access via Cisco Expressway Deployment Guide Cisco Expressway X8.1.1 or later Cisco Unified CM 9.1(2)SU1 or later January 2015 Contents Mobile and remote access

More information

Troubleshooting SIP Trunks with Cisco CallManager 5.x and Later

Troubleshooting SIP Trunks with Cisco CallManager 5.x and Later Troubleshooting SIP Trunks with Cisco CallManager 5.x and Later Document ID: 98524 Contents Introduction Prerequisites Requirements Components Used Conventions SIP with CallManager 5.x Media Termination

More information

Application Notes Rev. 1.0 Last Updated: February 3, 2015

Application Notes Rev. 1.0 Last Updated: February 3, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v8.6 for Level 3 Voice Complete SM Deployments Application Notes Rev. 1.0 Last Updated: February 3, 2015 Contents 1 Document Overview...

More information

Application Note: Cisco Integration with Onsight Connect

Application Note: Cisco Integration with Onsight Connect Application Note: Cisco Integration with Onsight Connect Table of Contents Application Note:... 1 Cisco Integration with Onsight Connect... 3 Direct Onsight Device to Cisco Endpoint Calls... 3 Cisco Unified

More information

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6.

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6. 1 SIP Carriers 1.1 Telstra 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

Application Notes Rev. 1.0 Last Updated: January 9, 2015

Application Notes Rev. 1.0 Last Updated: January 9, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v9.1 for Level 3 Voice Complete SM SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: January 9, 2015 Contents 1 Document

More information

Cisco Collaboration with Microsoft Interoperability

Cisco Collaboration with Microsoft Interoperability Cisco Collaboration with Microsoft Interoperability Infrastructure Cheatsheet First Published: June 2016 Cisco Expressway X8.8 Cisco Unified Communications Manager 10.x or later Microsoft Lync Server 2010

More information

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction 640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction Course Introduction Module 01 - Overview of Cisco Unified Communications Solutions Understanding

More information

White paper. SIP An introduction

White paper. SIP An introduction White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary

More information

Unified Communications Mobile and Remote Access via Cisco VCS

Unified Communications Mobile and Remote Access via Cisco VCS Unified Communications Mobile and Remote Access via Cisco VCS Deployment Guide Cisco VCS X8.1.1 or later Cisco Unified CM 9.1(2)SU1 or later January 2015 Contents Mobile and remote access 4 Jabber client

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments

SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: April 10, 2015 Revision Date Revised By Comments 0.1 12/03/2015 Roman

More information

IM and Presence Service Network Setup

IM and Presence Service Network Setup Configuration changes and service restart notifications, page 1 DNS Domain Configuration, page 2 IM and Presence Service Default Domain Configuration, page 6 IM Address Configuration, page 7 Domain Management

More information

Integrating Citrix EasyCall Gateway with SwyxWare

Integrating Citrix EasyCall Gateway with SwyxWare Integrating Citrix EasyCall Gateway with SwyxWare The EasyCall Gateway has been tested for interoperability with Swyx SwyxWare, versions 6.12 and 6.20. These integration tests were done by using EasyCall

More information

Conference Bridge setup

Conference Bridge setup Conference Bridge setup This chapter provides information to configure conference bridges using Cisco Unified Communications Manager Administration. See the following for additional information: Conference

More information

Non-Cisco SIP phones setup

Non-Cisco SIP phones setup n-cisco SIP phones setup This appendix provides information about Configuring n-cisco Phones That Are Running SIP. About non-cisco SIP phone setup, page 1 Third-party SIP phone setup process, page 1 Different

More information

Polycom Unified Communications Deployment Guide for Cisco Environments

Polycom Unified Communications Deployment Guide for Cisco Environments Polycom Unified Communications Deployment Guide for Cisco Environments Wave 5 March 2012 3725-00010-001G Trademark Information Polycom, the Polycom Triangles logo, and the names and marks associated with

More information

Implementing Cisco IP Telephony & Video, Part 1

Implementing Cisco IP Telephony & Video, Part 1 Course Code: CI-CIPTV1 Vendor: Cisco Course Overview Duration: 5 RRP: 2,320 Implementing Cisco IP Telephony & Video, Part 1 Overview Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day

More information

Session Manager Overview. Seattle IAUG Chapter Meeting

Session Manager Overview. Seattle IAUG Chapter Meeting Session Manager Overview Seattle IAUG Chapter Meeting Agenda Session Manager continues to evolve.. Flexibility BYOD Soft Clients Endpoints SIPenablement 3 rd Party Adjuncts Centralized SIP Trunking Redundancy

More information

Cisco TelePresence Video Communication Server Basic Configuration (Control with Expressway)

Cisco TelePresence Video Communication Server Basic Configuration (Control with Expressway) Cisco TelePresence Video Communication Server Basic Configuration (Control with Expressway) Deployment Guide Cisco VCS X8.1 D14651.08 August 2014 Contents Introduction 4 Example network deployment 5 Network

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure Mitel UC360 SIP Phone and Mitel MCD for use with VidyoWay SIP CoE 13-4940-00228 NOTICE The information contained in this document is believed to be

More information

Cisco Expressway Basic Configuration

Cisco Expressway Basic Configuration Cisco Expressway Basic Configuration Deployment Guide Cisco Expressway X8.1 D15060.03 August 2014 Contents Introduction 4 Example network deployment 5 Network elements 6 Internal network elements 6 DMZ

More information

Implementing Cisco Collaboration Applications **Part of the CCNP Collaboration certification track**

Implementing Cisco Collaboration Applications **Part of the CCNP Collaboration certification track** Course: Duration: Price: $ 3,795.00 Learning Credits: 38 Certification: Implementing Cisco Collaboration Applications Implementing Cisco Collaboration Applications**Part of the CCNP Collaboration certification

More information

Session Initiation Protocol (SIP) The Emerging System in IP Telephony

Session Initiation Protocol (SIP) The Emerging System in IP Telephony Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia

More information

Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment

Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment Application Notes Rev 1.0 P/N 550-06690 Last Updated: October 26, 2015 Revision History Revision Date Revised

More information

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...

More information

Sametime Unified Telephony Lite Client:

Sametime Unified Telephony Lite Client: Sametime Version 8.5.2 From Zero to Hero Sametime Unified Telephony Lite Client: Configuring SIP trunks to third-party audio/video equipment Contents Edition Notice...4 1 Introduction...5 1.1 What is Sametime

More information

How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions

How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions Overview This document provides a reference for configuration of the Avaya IP Office to connect to Integra Telecom SIP

More information

Cisco TelePresence Authenticating Cisco VCS Accounts Using LDAP

Cisco TelePresence Authenticating Cisco VCS Accounts Using LDAP Cisco TelePresence Authenticating Cisco VCS Accounts Using LDAP Deployment Guide Cisco VCS X8.1 D14465.06 December 2013 Contents Introduction 3 Process summary 3 LDAP accessible authentication server configuration

More information

Cisco TelePresence Video Communication Server (Cisco VCS) IP Port Usage for Firewall Traversal. Cisco VCS X8.5 December 2014

Cisco TelePresence Video Communication Server (Cisco VCS) IP Port Usage for Firewall Traversal. Cisco VCS X8.5 December 2014 Cisco TelePresence Video Communication Server (Cisco VCS) IP Port Usage for Firewall Traversal Cisco VCS X8.5 December 2014 Contents: Cisco VCS IP port usage Which IP ports are used with Cisco VCS? Which

More information

Implementation notes on Integration of Avaya Aura Application Enablement Services with Microsoft Lync 2010 Server.

Implementation notes on Integration of Avaya Aura Application Enablement Services with Microsoft Lync 2010 Server. Implementation notes on Integration of Avaya Aura Application Enablement Services with Microsoft Lync 2010 Server. Introduction The Avaya Aura Application Enablement Services Integration for Microsoft

More information

BroadSoft Partner Configuration Guide

BroadSoft Partner Configuration Guide BroadSoft Partner Configuration Guide Microsoft Lync 2010 SIP Trunking August 2012 Document Version 1.6 9737 Washingtonian Blvd Suite 350 Gaithersburg, MD USA 20878 Tel +1 301.977.9440 WWW.BROADSOFT.COM

More information

End User Setup and Handling

End User Setup and Handling on IM and Presence Service, page 1 Authorization Policy Setup On IM and Presence Service, page 1 Bulk Rename User Contact IDs, page 4 Bulk Export User Contact Lists, page 5 Bulk Export Non-Presence Contact

More information

Session Border Controller

Session Border Controller CHAPTER 13 This chapter describes the level of support that Cisco ANA provides for (SBC), as follows: Technology Description, page 13-1 Information Model Objects (IMOs), page 13-2 Vendor-Specific Inventory

More information

Cisco TelePresence MCU 45X0, 53X0 and MCU MSE 8510

Cisco TelePresence MCU 45X0, 53X0 and MCU MSE 8510 Cisco TelePresence MCU 45X0, 53X0 and MCU MSE 8510 Cisco TelePresence Deployment Guide May 2012 D14962 Contents Contents Introduction... 5 Audience... 5 Scope... 5 Background... 6 MCU overview... 6 Conference

More information

Implementing Cisco Unified Communications Manager Part 1, Volume 1

Implementing Cisco Unified Communications Manager Part 1, Volume 1 Implementing Cisco Unified Communications Manager Part 1, Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your Training

More information

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy INTERACTIVE INTELLIGENCE AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy Version 1.7 9/2/2009 TABLE OF CONTENTS 1 AT&T... 5 1.1 Introduction... 5 1.2 Product Descriptions...

More information

Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led

Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led Course Description Implementing Cisco Collaboration Devices (CICD v1.0) is an extended hours 5-day course focusing on providing

More information

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.0 Abstract These Application

More information

EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens

EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens Nick Marly, Dominique Chantrain, Jurgen Hofkens Alcatel Francis Wellesplein 1 B-2018 Antwerp Belgium Key Theme T3 Tel : (+32) 3 240 7767 Fax : (+32) 3 240 8485 E-mail : Nick.Marly@alcatel.be Tel : (+32)

More information

TMS Phone Books Troubleshoot Guide

TMS Phone Books Troubleshoot Guide TMS Phone Books Troubleshoot Guide Document ID: 118705 Contributed by Adam Wamsley and Magnus Ohm, Cisco TAC Engineers. Jan 05, 2015 Contents Introduction Prerequisites Requirements Components Used Related

More information

Troubleshooting Cisco Unified Communications (TVOICE)

Troubleshooting Cisco Unified Communications (TVOICE) Troubleshooting Cisco Unified Communications (TVOICE) Course Overview: Troubleshooting Cisco Unified Communications (TVOICE) prepares network professionals with the knowledge and skills that are required

More information

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Time Warner ITSP Setup Guide

Time Warner ITSP Setup Guide October 14 Time Warner ITSP Setup Guide Author: Zultys Technical Support This configuration guide was created to assist knowledgeable vendors with configuring the Zultys MX Phone System with Time Warner

More information

MODELLING OF INTELLIGENCE IN INTERNET TELEPHONE SYSTEM

MODELLING OF INTELLIGENCE IN INTERNET TELEPHONE SYSTEM MODELLING OF INTELLIGENCE IN INTERNET TELEPHONE SYSTEM Evelina Nicolova Pencheva, Vessela Liubomirova Georgieva Department of telecommunications, Technical University of Sofia, 7 Kliment Ohridski St.,

More information

Application Notes for the Ingate SIParator with Avaya Converged Communication Server (CCS) - Issue 1.0

Application Notes for the Ingate SIParator with Avaya Converged Communication Server (CCS) - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for the Ingate SIParator with Avaya Converged Communication Server (CCS) - Issue 1.0 Abstract These Application Notes describe the configuration

More information

This presentation discusses the new support for the session initiation protocol in WebSphere Application Server V6.1.

This presentation discusses the new support for the session initiation protocol in WebSphere Application Server V6.1. This presentation discusses the new support for the session initiation protocol in WebSphere Application Server V6.1. WASv61_SIP_overview.ppt Page 1 of 27 This presentation will provide an overview of

More information

Cisco Unified Communications Manager 7.0

Cisco Unified Communications Manager 7.0 Cisco Unified Communications Manager 7.0 Cisco Unified Communications Solutions unify voice, video, data, and mobile applications on fixed and mobile networks, enabling easy collaboration every time from

More information

Optional VBP-E at the Headquarters Location

Optional VBP-E at the Headquarters Location publicly whitelist/blacklist LAN/Subscriber-side GK address. Submit Default alias Optional VBP-E at the Headquarters Location As shown in the diagram above, you can choose to install a VBP-E to allow your

More information

Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013

Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013 Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013 This is to cover the steps needed for basic functionality to communicate with Etherspeak s SIP trunking service. Many environments are different

More information

Chapter 2 PSTN and VoIP Services Context

Chapter 2 PSTN and VoIP Services Context Chapter 2 PSTN and VoIP Services Context 2.1 SS7 and PSTN Services Context 2.1.1 PSTN Architecture During the 1990s, the telecommunication industries provided various PSTN services to the subscribers using

More information

Http://www.passcert.com

Http://www.passcert.com Http://www.passcert.com Exam : 70-337 Title : Enterprise Voice & Online Services with Microsoft Lync Server 2013 Version : DEMO 1 / 18 Topic 1, Litware, Inc Case A Overview Litware, Inc., is an international

More information

210-060. Implementing Cisco Collaboration Devices v1.0. Version: Demo. Page <<1/10>>

210-060. Implementing Cisco Collaboration Devices v1.0. Version: Demo. Page <<1/10>> 210-060 Implementing Cisco Collaboration Devices v1.0 Version: Demo Page 1. Which two technologies comprise a Cisco Presence deployment? (Choose two.) A. Cisco Unified Presence Server B. Cisco

More information

Configuring a Pure-IP SIP Trunk in Lync 2013

Configuring a Pure-IP SIP Trunk in Lync 2013 Configuring a Pure-IP SIP Trunk in Lync 2013 Contents Configuring a Pure-IP SIP Trunk in Lync 2013... 1 Introduction - Product version: Microsoft Lync Server 2013... 2 Pure-IP SIP Trunk configuration tasks...

More information

Integrating VoIP Phones and IP PBX s with VidyoGateway

Integrating VoIP Phones and IP PBX s with VidyoGateway Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES

More information

Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options

Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Document Summary This document provides information on several integration scenarios

More information

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course 1.Course Title Version dated 25/11/2014 NATO Voice over IP Foundation Course 2.Identification Number (ID) 095 3. Purpose of the Course There are a number of new technologies (to NATO) that are encompassed

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIP CoE Technical Configuration Notes Configure MCD for use with OpenIP SIP Trunking service SIP CoE 11-4940-00186 NOTICE The information contained in this document is believed to be accurate in

More information

Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0

Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0 Avaya Solution & Interoperability Test Lab Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0 Abstract These Application Notes describe

More information

1 SIP Carriers. 1.1.1 Warnings. 1.1.2 Vendor Contact Vendor Web Site : http://www.wind.it. 1.1.3 Versions Verified SIP Carrier status as of 9/11/2011

1 SIP Carriers. 1.1.1 Warnings. 1.1.2 Vendor Contact Vendor Web Site : http://www.wind.it. 1.1.3 Versions Verified SIP Carrier status as of 9/11/2011 1 SIP Carriers 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found in the SIP

More information

Feature and Technical

Feature and Technical BlackBerry Mobile Voice System for SIP Gateways and the Avaya Aura Session Manager Version: 5.3 Feature and Technical Overview Published: 2013-06-19 SWD-20130619135120555 Contents 1 Overview...4 2 Features...5

More information

Configuration Notes 283

Configuration Notes 283 Mediatrix 4400 Digital Gateway VoIP Trunking with a Legacy PBX June 21, 2011 Proprietary 2011 Media5 Corporation Table of Contents Table of Contents... 2 Introduction... 3 Mediatrix 4400 Digital Gateway

More information

Application Notes for Microsoft Office Communicator Clients with Avaya Communication Manager Phones - Issue 1.1

Application Notes for Microsoft Office Communicator Clients with Avaya Communication Manager Phones - Issue 1.1 Avaya Solution & Interoperability Test Lab Application Notes for Microsoft Office Communicator Clients with Avaya Communication Manager Phones - Issue 1.1 Abstract These Application Notes describe the

More information

Configuring SIP Trunking and Networking for the NetVanta 7000 Series

Configuring SIP Trunking and Networking for the NetVanta 7000 Series 61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking

More information

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313 MITEL SIP CoE Technical Configuration Note Configure MCD for use with Intelepeer Service provider SIP Trunking SIP CoE 14-4940-00313 NOTICE The information contained in this document is believed to be

More information

Understand SIP trunk and registration in DWG gateway Version: 1.0 Dinstar Technologies Co., Ltd. Date: 2014. 09.29

Understand SIP trunk and registration in DWG gateway Version: 1.0 Dinstar Technologies Co., Ltd. Date: 2014. 09.29 Understand SIP trunk and registration in DWG gateway Version: 1.0 Dinstar Technologies Co., Ltd. Date: 2014. 09.29 http://www.dinstar.com 1 / 9 Contents Chapter 1: Authors and changes logs... 3 Chapter

More information

IP Office 7.0 and BCM 6.0 SIP Interoperability Configuration Notes

IP Office 7.0 and BCM 6.0 SIP Interoperability Configuration Notes IP Office 7.0 and BCM 6.0 SIP Interoperability Configuration Notes Abstract: This document provides information on how to configure a network solution with IP Office 7.0 and BCM 6.0 using SIP trunks. 2011

More information

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197 MITEL SIP CoE Technical Configuration Note Configure MCD for use with SIP Trunking Service SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide EarthLink Business SIP Trunking Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide Publication History First Release: Version 1.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure Inn-Phone SIP Phone for use with MCD SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

Need for Signaling and Call Control

Need for Signaling and Call Control Need for Signaling and Call Control VoIP Signaling In a traditional voice network, call establishment, progress, and termination are managed by interpreting and propagating signals. Transporting voice

More information

Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0

Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0 Abstract These Application Notes describe the steps

More information

Key Elements of a Successful SIP Device Provisioning System

Key Elements of a Successful SIP Device Provisioning System Key Elements of a Successful SIP Device Provisioning System A white paper by Incognito Software April, 2006 2006 Incognito Software Inc. All rights reserved. Page 1 of 6 Key Elements of a Successful SIP

More information

Using LifeSize Systems with Microsoft Office Communications Server 2007

Using LifeSize Systems with Microsoft Office Communications Server 2007 Using LifeSize Systems with Microsoft Office Communications Server 2007 This technical note describes the steps to integrate a LifeSize video communications device with Microsoft Office Communication Server

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

BROADSOFT PARTNER CONFIGURATION GUIDE VEGASTREAM VEGA 100

BROADSOFT PARTNER CONFIGURATION GUIDE VEGASTREAM VEGA 100 BROADSOFT PARTNER CONFIGURATION GUIDE VEGASTREAM VEGA 100 JULY 2005 Version 1.0 BroadWorks Guide Copyright Notice Copyright 2005 BroadSoft, Inc. All rights reserved. Any technical documentation that is

More information